Re: [Sursound] A comparison of fifteen ambisonic microphones

[eric benjamin]

Where did that facebook discussion appear? I'm interested in the mechanism
by which microphone comparisons are made.

On Thu, Nov 2, 2023 at 12:24 PM  wrote:

> We discussed the multiple errors in detail in our Facebook discussion.
>
> That you didn't correct the comparison study, and actually added more
> incorrect information has made it clear
> that further discussion won't improve the outcome.
>
>
> Len Moskowitz (mosko...@core-sound.com)
> Core Sound LLC
> www.core-sound.com
> Home of OctoMic and TetraMic
>
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Re: [Sursound] Immersive Installation Kew

[eric benjamin]

Unfortunately, your attachment didn't arrive. Perhaps you could post it
somewhere and send a link.

I'd love to experience your installation but I'm 7,000 miles away. We do
have a lot of lovely venues in San Francisco if you decide to take it on
the road!

On Tue, Oct 18, 2022 at 3:35 AM Augustine Leudar 
wrote:

> Dear all,
> If anyone's in London, I have a sound installation running at Kew Gardens
> in the temperate house till the end of October. Its been through several
> incarnations and different speaker arrays, the current is a bit unusual in
> that its two lines of speakers one at height, one at ground level 60m long,
> so sounds travel up and down one Edge of the building
>
> Link here :
>
> https://www.kew.org/read-and-watch/mexico-magico
>
> I attach a document for this list which details Artistic, Scientific and
> Technical information for anyone interested
> Al the best,
> Gus
>
> --
> Artist website: www.augustineleudar.com
> Business website: www.magikdoor.net
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Re: [Sursound] A 7th-order array with 16 microphones

[eric benjamin]

I believe that Nando may have been thinking about reproduction with
loudspeaker arrays. He has a system with eight loudspeakers on the
horizontal plane, as do I. So good up to third order. But I actually have
24 full-range loudspeakers available. Would it be advantageous to expand
our systems to higher order? I've been asking this question for a very long
time. With higher order program material slowly becoming available, perhaps
we can find out.

Eric Benjamin

On Thu, Dec 2, 2021 at 5:57 AM Jens Ahrens  wrote:

> Hi Fons, hi Nando,
>
> Please excuse that I’m responding to both of you in the same mail. There
> is sufficient overlap in the matters to keep the thread from diverging.
>
> @Fons: Thanks for the clarification! We will look into this.
>
> @Nando: (The question was what the high orders contribute.)
>
> It’s hard to tell how exactly the high orders contribute. One aspect is
> the interaural coherence that needs to be appropriate. The other main
> aspect is what I typically term the equalization: Below the aliasing
> frequency, things are fine anyway. Above the aliasing frequency, the
> spectral balance of the binaural signals tends to be more even the higher
> the orders are that are present. The deviations from the ideal spectral
> balance also tend to be less strongly dependent on the incidence angle of
> the sound if higher orders are present.
>
> Much of the angle dependent deviations of the spectral balance can be
> mitigated, for example, by MagLS so that the perceptual difference between,
> say, 7th order and infinite order is small. I can’t tell if it gets any
> smaller with higher orders. My (informal) feeling is that somewhere between
> 5th and 10th order is where the perceptual difference to the ground truth
> saturates, both in terms of equalization and the coherence.
>
> Best regards,
> Jens
>
>
>
> > On 2 Dec 2021, at 11:20, Fons Adriaensen  wrote:
> >
> > Hi Jens,
> >
> >> I’m attaching Fig. 1 from the JASA article.
> >
> > Nothing was attached (or it got lost...)
> >
> >> If I’m not misreading, then the 7th order is available somewhere between
> >> 2 kHz and 3 kHz and higher. Aliasing kicks in at around 4 kHz-ish.
> >
> > So the question is if this small range (less than one octave) actually
> > contributes anything useful.
> >
> >> My guess is that it is not more or less sensitive than SMAs.
> >
> > I'd agree.
> >
> >> I’m as close as a few centimetres to the surface of the array. This
> >> triggers a lot of the high orders at low frequencies, and if there
> >> is something that is not ideal, then the low frequencies tend to go
> >> through the ceiling.
> >
> > If they don't that could just be because their contribution at LF
> > is filtered out anyway, e.g. if your A/B process includes high pass
> > filters of an order at least one higher than the order of the
> > component they act on.
> >
> >> How would I be noticing if the microphone mismatch is above
> >> the tolerance level?
> >
> > One way uncalibrated capsule gains will show up is that after
> > binaural rendering you get significant ILD at LF, which should
> > never happen except for very close sources.
> >
> > This actually happened recently with a binaural rendering system
> > I was working on. When the room sound (early reflections and
> > reverb tail) was added, this resulted in excessive ILD at LF,
> > and a perception of the room sound that was clearly biased to
> > one side.
> >
> > The room sound in this case was from a real room, measured using
> > an SMA. Analysing these measurements revealed capsule gain errors
> > up to +/-3 dB. When these were compensated for, the problem
> > disappeared.
> >
> >
> > You could just measure the B-format polars at LF, but that would
> > require an anechoic room.
> >
> > You could instead compute the theoretical capsule signals for a
> > set of directions, apply some gain errors, send the result through
> > your A/B process, and plot the result.
> >
> > The only thing that mitigates this problem is statistics: with
> > a high number of capsules contributing to each harmonic, errors
> > tend to average out to some extent.
> >
> > Ciao,
> >
> > --
> > FA
> >
> > ___
> > Sursound mailing list
> > Sursound@music.vt.edu
> > https://mail.music.vt.edu/mailman/listinfo/sursound - unsubscribe here,
> edit account or options, view archives and so on.
>
> ___
> Sursound mail

Re: [Sursound] Fwd: audeze

[Eric Benjamin]

Way too big.

Sent from Mail for Windows 10

From: seva, soundcurrent mastering
Sent: Monday, December 10, 2018 9:20 PM
To: Surround Sound discussion group
Subject: [Sursound] Fwd: audeze

hi all round,

yes, a joke there.

anyone have a view on *audeze tetrahedral*? seems like with a 100mm
diaphragm the inter-capsule distance is too big.



-- 
*seva*
Chief Engineer Sequoyah Studios
recording - mixing - mastering - archiving - forensic - post
sequoyahstudios.com
linkedin profile 
portfolio 



-- 
best
seva

s...@soundcurrent.com
www.soundcurrent.com
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Re: [Sursound] OctoMic VR 8k Video, New Files on OctoMic Recording Download Page

[Eric Benjamin]

I’d love to hear the second order version.

Sent from Mail for Windows 10

From: Len Moskowitz
Sent: Tuesday, November 6, 2018 9:08 AM
To: Sursound List
Subject: [Sursound] OctoMic VR 8k Video,New Files on OctoMic Recording Download 
Page

Phillip Westbrook posted on Facebook:

  "A recent production done in Venice Italy with the OctoMic! The second 
order version of course sounds better, but I'm really enjoying how easy 
the OctoMic is to use."


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


[YouTube videos currently offer only first-audio ambisonic audio.]


-


In addition, Core Sound posted a few new files to the OctoMic Recording 
Download web page:


   www.core-sound.com/OctoMic/13.php


Len Moskowitz (mosko...@core-sound.com)
Core Sound LLC
www.core-sound.com
Home of OctoMic and TetraMic

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Re: [Sursound] RIR measuring, how to capture a higher order Ambisonic room responce?

[Eric Benjamin]

For measurement purposes it should be possible to generate higher order 
responses using two or even one figure eight microphone. Place two figure eight 
microphones facing in the same direction spaced about one inch or so apart, 
depending on the frequency range desired. This could also be done by using a 
single figure eight microphone. Measure the impulse response in one direction, 
and then move the microphone about one inch away and measure another impulse 
response.  Subtract the two IRs and obtain a higher order IR. This is more 
difficult than my simple explanation. I haven’t done this, so caveat emptor.

Eric Benjamin.




From: Bo-Erik Sandholm
Sent: Tuesday, April 24, 2018 8:58 AM
To: sursound
Subject: Re: [Sursound] RIR measuring,how to capture a higher order Ambisonic 
room responce?

Thank you all, for your answers.
I received a lot of information.

I will start with FOA RIRs from my tetramic , it seems my ideas for
measuring higher orders are not realistic without a higher order microphone.

And FOA is probably good enough for my proof of concept.



Best Regards
Bo-Erik Sandholm
Stockholm

On Tue, 24 Apr 2018 03:59 umashankar manthravadi, <umasha...@hotmail.com>
wrote:

> I have been using a stepper motor (of the kind used in 3d printer ) driven
> by a low cost Arduino and motor control board. I 3d print a snug fitting
> fixture for the microphone with the motor shaft  aligned to the array
> centre. It is low cost so I design a fitting for each mic I test, including
> the Brahma-in-Zoom. A small Arduino script rotates the stepper 25 steps
> each time I press a button (for 16 positions) and 50 steps (for 8
> positions). I was worried about the stepper skipping with the weight of the
> microphone, but that is not happening, even with a five volt supply. I was
> ready with a thrust bearing between the motor housing and the microphone
> housing but it was not necessary. I plan to get rid of the switch and use a
> pulse on the right channel instead, though I generally do not like to
> automate things too much.
>
>
>
> umashankar
>
>
>
> Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for
> Windows 10
>
>
>
> 
> From: Sursound <sursound-boun...@music.vt.edu> on behalf of Fernando
> Lopez-Lezcano <na...@ccrma.stanford.edu>
> Sent: Tuesday, April 24, 2018 1:40:56 AM
> To: Surround Sound discussion group
> Subject: Re: [Sursound] RIR measuring, how to capture a higher order
> Ambisonic room responce?
>
> On 04/23/2018 12:42 PM, Stefan Schreiber wrote:
> >> I can do the 4 measurements with 45 degrees rotation of my tetramic,
> that
> >> is not so difficult,  the next step to create a second order ambisonic
> >> RIR
> >>
> >> that is where I will fail :-).
>
> You would need to "calibrate" the created 8 capsule array. That is,
> record impulse responses all around it in a big space or anechoic room
> (enough to accurately sample the spherical harmonics you want), and then
> derive an A to B converter from that. I have some preliminary code in my
> SpHEAR project that tries to do that, but it is not a "push a button and
> you are done" thing at all...
>
> For Fons's code, and to do this the "right way"...
> On 03/27/2018 01:18 PM, Fons Adriaensen wrote:
> > ...  you'll have to sell your soul :-)
>
> :-P
>
> > I believe you might need a quite high precision to be successful even at
> > the first step...
> >
> > (A SF mike has narrowly spaced capsules, and needs calibrationThe
> > mechanical precision you need to measure 2nd order with a FOA mike is
> > IMHO high.)
>
> Based on my experience with the Octathingy's I have built I would agree,
> you would need to be very precise (and repeatable).
>
> In my case to get good calibration data I need to rotate the microphone
> with no wobble and at different orientations (or if it is not _exactly_
> perfect, try to get away with calibrating out the average delays to all
> capsules).
>
> BTW, I cannot move the speaker around which would probably be a better
> solution because of space constraints... I can barely get 4.5mSecs of IR
> data in the spaces I can use.
>
> > So the mathematical methods (based on FOA but improving the RIR
> > resolution, as suggested by Archontis) should be a better way to go
> > on... Especially since you could receive even higher resolutions/orders,
> > and in practice.
> >
> > So the presented ideas to capture 2nd order RIRs via a 1st order mike
> > are brilliant, but are they practical?
>
> Probably not practical IMHO.
>
> > And even if somebody could succeed in a very careful process: this does
> > no

Re: [Sursound] RIR measuring, how to capture a higher order Ambisonic room responce?

[Eric Benjamin]

I did this rotation and calibration operation. Unfortunately the results were 
not great. When the array is rotated it has to overlay the previous position 
perfectly.  There is also a tendency for the mic stand to wobble when it 
rotates. These results are shown in my AES paper on the second order microphone.

Eric Benjamin

From: Fernando Lopez-Lezcano
Sent: Monday, April 23, 2018 1:11 PM
To: Surround Sound discussion group
Subject: Re: [Sursound] RIR measuring,how to capture a higher order Ambisonic 
room responce?

On 04/23/2018 12:42 PM, Stefan Schreiber wrote:
>> I can do the 4 measurements with 45 degrees rotation of my tetramic, that
>> is not so difficult,  the next step to create a second order ambisonic
>> RIR
>>
>> that is where I will fail :-).

You would need to "calibrate" the created 8 capsule array. That is, 
record impulse responses all around it in a big space or anechoic room 
(enough to accurately sample the spherical harmonics you want), and then 
derive an A to B converter from that. I have some preliminary code in my 
SpHEAR project that tries to do that, but it is not a "push a button and 
you are done" thing at all...

For Fons's code, and to do this the "right way"...
On 03/27/2018 01:18 PM, Fons Adriaensen wrote:
> ...  you'll have to sell your soul :-)

:-P

> I believe you might need a quite high precision to be successful even at
> the first step...
>
> (A SF mike has narrowly spaced capsules, and needs calibrationThe
> mechanical precision you need to measure 2nd order with a FOA mike is
> IMHO high.)

Based on my experience with the Octathingy's I have built I would agree, 
you would need to be very precise (and repeatable).

In my case to get good calibration data I need to rotate the microphone 
with no wobble and at different orientations (or if it is not _exactly_ 
perfect, try to get away with calibrating out the average delays to all 
capsules).

BTW, I cannot move the speaker around which would probably be a better 
solution because of space constraints... I can barely get 4.5mSecs of IR 
data in the spaces I can use.

> So the mathematical methods (based on FOA but improving the RIR
> resolution, as suggested by Archontis) should be a better way to go
> on... Especially since you could receive even higher resolutions/orders,
> and in practice.
>
> So the presented ideas to capture 2nd order RIRs via a 1st order mike
> are brilliant, but are they practical?

Probably not practical IMHO.

> And even if somebody could succeed in a very careful process: this does
> not look to be a robust measurement method. ..
>
> We always talk about the 1st reflections, in this case. Not reverb,
> which is kind of statistical.
>
> Of course you can try, but how much precision is really needed? (Should
> be clarified before...)

I would have to go to my data to get some numbers... I definitely can 
see effects at high frequencies when the data capture is not precise 
(I'm in the process of trying to build a better measuring rig).

-- Fernando

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Re: [Sursound] Core Sound Announces OctoMic - First 2nd-orderAmbisonics Microphone

[Eric Benjamin]

I’d like to point out that I am the inventor of the second order microphone 
array called by Core-Sound the Octomic. I published the design six years ago at 
the 133rd AES Convention in preprint 8728. I purposefully did not patent the 
invention to allow entrepreneurs to use the design.  But I would like a little 
bit of credit! I realize that Core Sound put in significant effort in making 
their product.

I made the microphone utilize only eight channels because I only can record 
eight channels.

Eric Benjamin

From: Jörn Nettingsmeier
Sent: Tuesday, March 27, 2018 12:09 PM
To: sursound@music.vt.edu
Subject: Re: [Sursound] Core Sound Announces OctoMic - First 
2nd-orderAmbisonics Microphone

On 03/27/2018 03:59 PM, Len Moskowitz wrote:
> core-sound.com/OctoMic/1.php

Sweet! A resounding "me too" to Stefan's question about the matrix, 
since you're one channel short :)
Looking at the geometry, I guess you sacrificed the second-order 
rotationally symmetric component (FuMa R or ACN 08), which seems to be a 
good choice to me.

What's that disclaimer about third-party PPAs? I mean, your PPA seems to 
be an integral part of the mic, given that it produces unbalanced 
signals. Is there any other magic going on there?

I wonder if you can get ZOOM to include an octomic firmware in the F8 
eventually, as Sennheiser did with their Ambeo mic, that one's mighty 
handy. (Not that I'm holding my breath, I wouldn't be surprised if there 
were some exclusive deals at play...)

Can I assume that there is an updated version of TetraProc for the 
followers of the penguin?

And looking at the shop, I don't see the Rycote lyre listed yet - is it 
a generic one that can be had from them, or something custom-made and 
not quite ready yet?

I'm trying to come up with a good business case to order one asap, and 
since I'm good at fooling myself, it might just happen :-D


All best,


Jörn




-- 
Jörn Nettingsmeier
Tuinbouwstraat 180, 1097 ZB Amsterdam, Nederland
Tel. +49 177 7937487

Meister für Veranstaltungstechnik (Bühne/Studio), Tonmeister VDT
http://stackingdwarves.net
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Re: [Sursound] Sounds in Space 2018

[Eric Benjamin]

Once again, congratulations! I wish I could be there. I’m on the wrong 
continent.

From: Bruce Wiggins
Sent: Thursday, February 22, 2018 8:42 AM
To: Surround Sound discussion group
Subject: [Sursound] Sounds in Space 2018

I'm pleased to announce that this year's Sounds in Space Research
Symposium on Spatial Audio will be a two day event (26th/27th June in
Derby, UK). The call for works is now open looking at both the
technical and artistic side of spatial audio. We'll have a 28+ 3D
speaker array in the auditorium for demos/talks and we've got videos
of last year's event if you'd like to get a feel for what goes on.

We're also looking for sponsors/exhibitors at the event, too.

http://soundsinspace.co.uk/
(last year's event videos/presentations :
https://www.brucewiggins.co.uk/?page_id=881)

cheers

Bruce Wiggins
University of Derby
UK
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[Sursound] ambisonic recording of the philharmonic orchestra

[Eric Benjamin]

https://www.philharmonia.co.uk/digital/virtual_reality_and_apps/mahler_3

I’m looking forward to this. Does any one have further details?

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Re: [Sursound] Live ambisonic electronica in London

[Eric Benjamin]

I’d love to come but I live near San Francisco. If you should choose to show 
again near hear I’ll definitely attend.

Sent from Mail for Windows 10

From: Tom Slater
Sent: Tuesday, November 14, 2017 7:41 AM
To: sursound@music.vt.edu
Subject: [Sursound] Live ambisonic electronica in London

Hello,

*Come and see Warsnare 360!*

Call & Response are producing a 29-loudspeaker ambisonic gig


*24th November 7.30 till late*
*Albany Theatre, Douglas Way London SE8 4AG*

*Get tickets here
*

I've been working with electronica artist Warsnare at our ambisonic studio
to mix his stereo album into a live 3D audio extravaganza.

His stuff is a mixture of urban music styles; Drum & Bass, House, Hip Hop.
We've got Kate Tempest on one of the tracks, support acts from Goldsmiths
NX Records and even a surround sound DJ set to finish off the night.

For the gig I'll be installing a dome of  29 loudspeakers with a 10m radius
that surrounds the audience and I'll be moving the audio from live string
players, percussion, synths and vocals around the space in real time.

*Work in progress*
Have a look at these 360 teaser vids of our studio sessions

https://www.youtube.com/watch?v=hnUz94qyoEw
https://www.youtube.com/watch?v=NOCYGs1tZ-8

I hope you can make it down, please share the details with anyone you think
will be interested.

Thanks,

Tom
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Re: [Sursound] wireless speaker/transmission recommendations

[Eric Benjamin]

Jim
You might want to give us a target budget for this distribution system.

Sent from Yahoo Mail on Android 
 
  On Wed, Nov 16, 2016 at 5:24 PM, Wim wrote:   Hi Jim,

You could have a look at the Neutrik Xirium system. It's the only one I
know that is capable of timed multi channel operation:

http://www.neutrik.com/en/news/xirium-digital-wireless-audio

It is expensive, as is to be expected with this number of channels.

Cheers,


Wim

2016-11-16 23:06 GMT+01:00 Augustine Leudar :

> Helmut - I;m not sure thats what he's looking for but it looks cool - is it
> wavefield synthesis system ? How to you do the vertical panning ? Do you
> use amplitude panning for vertical between horizontal WFS arrays like Matt
> Motags WFS designer ?
>
> http://www.mattmontag.com/projects-page/wfs-designer
>
> would love to hear it - let me knoe if you demo in the UK or Ireland
>
>
> On 16 November 2016 at 21:33, Helmut Oellers  wrote:
>
> > ...Show www.holoplot.com or www.syntheticwave.de
> >
> > H.
> >
> > Am 16.11.2016 21:57 schrieb "jim moses" :
> >
> > > Hi,
> > > I'm looking for suggestions for an installation project. I need to
> > transmit
> > > multi-channel audio to up to 10 loudspeakers. The distances are pretty
> > > short but I'm looking for a system with flexible routing.
> > >
> > > Thanks,
> > > Jim
> > >
> > > --
> > > Jim Moses
> > > Technical Director/Lecturer
> > > Brown University Music Department and M.E.M.E. (Multimedia and
> Electronic
> > > Music Experiments)
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> Augustine Leudar
> Artistic Director Magik Door LTD
> Company Number : NI635217
> Registered 63 Ballycoan rd,
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Re: [Sursound] Reduced set of B-format (or HOA) to binaural filters

[Eric Benjamin]

I have used that symmetry in the past. I assume that everyone does. 

Sent from Yahoo Mail on Android 
 
  On Wed, May 25, 2016 at 8:37 AM, Politis 
Archontis wrote:   Hi,

There has been some discussion before on B-format or HOA to binaural filters 
(HRTF-based), which no matter if it goes through a virtual decoder or with a 
direct HRTF-to-Bformat approach, two times the number of HOA channels of 
filters are needed (so 8 for B-format, 18 for 2nd-order HOA, 32 for 3rd-order 
HOA etc. ).

Playing around a bit I realized that if the HRTFs are made left-right 
symmetric, which makes sense to always force for non-individialized ones (and 
also for individualized sometimes), then a smaller set of filters is needed due 
to this left-right symmetry. For example for B-format binaural filters, the W, 
X and Z channel would have exactly the same filters, while the Y channel would 
have the same filter with a polarity inversion for one ear with respect to the 
other. Hence 4 filters instead of 8.

Thinking about it, it simply makes sense, as the spherical harmonics are 
themselves either symmetric or antisymmetric with respect to the x-z (median) 
plane. This is also what makes mirroring of a HOA sound scene from left to 
right for example, as easy as inverting certain HOA channels.

My question to any of the decoder developers on the list is if you have seen 
that anywhere analyzed or mentioned in the context of binaural decoding??

Many thanks,
Archontis Politis





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Re: [Sursound] Zoom H2n firmware update for spatial audio + surround podcast produciton

[Eric Benjamin]

The principle problem with using the ZoomH2n for Ambisonics recording is that 
the capsules are arranged in such a way that left obstructs right and right 
obstructs left. This can be seen in the photos on Daniel's web page 
athttp://www.radio.uqam.ca/ambisonic/images/h2_01.jpg
The result of the obstruction is that the polar patterns are severe!y 
compromised. They can't be modeled as a simple sum of zero and first order 
components which means that the recovered B-format will have significant 
errors. Also , the directions of the polar patterns vary with frequency. 
This doesnt mean that it shouldn't be used, just that it could be better. Any 
one who is willing to modify could try exchanging the positions of the capsules 
between left and right. Or changing them to a square configuration. 

Sent from Yahoo Mail on Android 
 
  On Wed, May 18, 2016 at 2:06 PM, Augustine Leudar 
wrote:   I guess it would be able to position sounds on the horizontal plane, 
the
gap between the capsules means high frequencies with wavelengths shorter
than that gap might be a bit off . I do a fair bit of sound design - my
opinion is that for stereo theres not much point in trying to create
"immersive 3d sound" - Id say a minimum of quadrophonic would be necesary.
Of course broacasting surround sound is a great option. Theres loads more
that you can do than just ambisonics though .

On 18 May 2016 at 17:40, Courville, Daniel  wrote:

> Brendan Baker wrote:
>
> >this update now enables the H2N to spit out a 4-channel WAV file somewhat
> resembling b-format, minus the vertical channel.
>
> It is B-Format, ambiX: ACN channel sequence (WYZX), SN3D normalization.
>
> >My assumption was/is that the orientation of the H2N's mic capsules
> wouldn't create a true ambisonic image.
>
> Why?
>
> Here's a small recording I did yesterday after updating the Zoom H2n. It's
> a street corner in Montreal, just beside UQAM.
>
> http://www.radio.uqam.ca/ambisonic/audio/mtl-uqam-acn-sn3d.flac
>
> FuMa version: http://www.radio.uqam.ca/ambisonic/audio/mtl-uqam-fuma.flac
>
> - Daniel
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-- 
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Artistic Director Magik Door LTD
Company Number : NI635217
Registered 63 Ballycoan rd,
Belfast BT88LL
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Re: [Sursound] Small multichannel speakers setup

[Eric Benjamin]

This is all good stuff. I'd like to add some info about the system aspects.

Speaker mounting. This can be nontrivial. I know of one professional periphonic 
installation where the cost of the stands was more than the speakers. And 
mounting speakers at height can be very difficult. If the speakers are heavy 
then it can be difficult and dangerous. I'm right in the middle of mounting 
several 50 lb speakers on the ceiling of my listening room. I couldn't find a 
suitable manufactured mount so I'm fabricating my own. My felling is plywood 
just to be able to support the speakers. 

Passive vs powered
When I built my first system there were very few inexpensive multichannel power 
amplifiers, so powers speakers seamed like the way to go. But powered speakers 
require that you run two cables instead of just one to each speaker. I'm still 
doing that. Now there are (relatively) inexpensive amplifiers like the Dayton 
1240 that give lots of channels. And Marc's suggestion of very inexpensive 
amplifier modules is tempting. Too bad no one makes something like 8 x 15 watts 
in a 1u package.

Coaxial speakers vs conventional 2-way
I'm including full-range in with coax. Enthusiasts of coax tout the fact that 
coax speakers are effectively point sources. My belief is that the primary 
advantage is in the off-axis responses. Two way speakers are designed to have 
flat on-axis response, but off axis there are nulls in the response which 
appear above and below the principle axis. But there are relatively few coaxial 
speakers from which to choose. 

That's it for now. 

Sent from my iPhone

> On May 13, 2016, at 10:25 AM, Marc Lavallee  wrote:
> 
> On Fri, 13 May 2016 09:44:58 -0700, mgra...@mstvp.com wrote:
>>> From: "Emanuele Costantini":
>>> Gallo A'Diva and ORB audio, I think they need an amplfier in order
>>> to work, which is not an option due to lack of room space.
>> 
>> I would bet that the spherical speakers and the requisite amplifier
>> take up not more space than most powered speakers. 
>> If you're going cost conscious you can use the small digital amps
>> from the likes of SMSL & Parts Express. I have one that delivers
>> 50w/c in 1/4 of 1 RU. That is, you can fit 4 into a 1 RU space.
> 
> I use a few identical amp modules, with 2 or more channels per amp:
> http://store3.sure-electronics.com/audio/audio-amplifier-board?dir=asc=price
> In a small room, with small speakers, 15 watts per channel is enough.
> The same store have a selection of power supplies. My solution is to
> use the power supply of a dedicated PC, where I installed all the amp
> modules in the drive bays. The 450W power supply is completely silent,
> as well as the rest of the PC (no fans at all, only passive cooling).
> --
> Marc
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Re: [Sursound] YouTube now supports Ambisonics (warning....part advertisement..)

[Eric Benjamin]

When I recorded Haley Jo's lovely voice for the original eight directions, I 
also recorded her announcing other directions. Unfortunately I wasn't prescient 
enough to record every conceivable direction. I wonder what would be a good 
announcement for half height positions?

Sent from my iPhone

On May 10, 2016, at 12:20 PM, Michael Chapman  wrote:

>> Aaron Heller wrote:
>> 
>>> Problem solved.  Here's the YouTube version of one of the most
> downloaded file from Ambisonia, AJH_eigtht_positions.amb
>>> 
>>>  https://youtu.be/eY9DMn8pgGA
> 
> 
>> Well done.
>> 
>> Just one suggestion: As VR is a lot about 3D, you could think about to
> extend the "8 positions" demo to some equivalent 3D demo. Which probably
> would be "26 positions". (3 rings w/ 8 positions each at -45�, 0�
> and 45� elevation. 2 additional positions at +-90�, i.e.
> zenith/nadir.)
>> 
>> Stefan
> 
> Happy to have mine used (with usual academic courtesies ...).
> (It's on ambisonia, or I can send.
> Am working on a HOA version ...)
> 
> Michael
> 
> 
> 
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Re: [Sursound] Rapture3D Universal SDK

[Eric Benjamin]

It's necessary to remove the period to make the link work.

Sent from Yahoo Mail on Android 
 
  On Thu, Mar 24, 2016 at 5:18 AM, Richard Furse wrote:   
[Warning: product announcement - I'll keep it brief.]

 

Hi there!

 

I thought a few folk here might be interested to know that Blue Ripple Sound
got around to releasing the Rapture3D Universal Software Development Kit
last week, see
http://www.blueripplesound.com/products/rapture3d-universal-sdk. This is a
portable realtime HOA renderer for gaming and Virtual Reality. There are C#
scripts for use in Unity (no OpenAL this time).

 

For linear content, you can use HOA mixes put together with our TOA VST
plugins (or others) as 3D "beds". These can make up some or all of the audio
scene and can rotate in response to a head-tracker or suchlike (e.g. in
Unity). For more conventional interactive gaming, there's also support for
large numbers of individually rendered mono objects, which can be combined
with beds as required before decoding to various formats (binaural, stereo,
5.1 etc.).

 

Best wishes,

 

--Richard

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Re: [Sursound] Furse-Malham to ACN conversion

[Eric Benjamin]

"In both cases the sound was coming from seemingly random places, and a
number of positions went practically silent."
What is needed, not just for you, but for everyone, is a comprehensive set of 
test files. It may be that your loudspeakers aren't where the system thinks 
they are (wrong speaker assignments), or it may be that that the decoder is 
doing the wrong thing. I have more extensive versions of test files, including 
"with height" like the eight directions files on Ambisonia., featuring the 
voice of the lovely Haley Jo. I could upload those if anyone would like them.
You can then use metering to determine if the specific sounds light up the 
speaker(s) that they should.

Sent from Yahoo Mail on Android 
 
  On Tue, Mar 22, 2016 at 1:18 PM, Jörn 
Nettingsmeier wrote:   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



-- 
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] The Mike Skeet Collection

[Eric Benjamin]

Through a strange coincidence, I emailed Mike last night to discuss one of his 
binaural recordings that I had just listened to. To read that he has passed on 
is eery. I was only acquainted with him, but the last time that we corresponded 
he invited me to drop by for a pint, not realizing that I was half the world 
away.

He will be remembered, for the legacy of his many recordings.

Eric Benjamin
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Re: [Sursound] SQ QUAD

[Eric Benjamin]

Richard,
You say "the two software programs you've been provided links for don't decode 
it" Just out of curiosity, in what way do the two software decoders fail to 
properly decode SQ? My interest is purely academic, as I don't have any SQ 
source material.
Eric Benjamin 


 On Tuesday, October 20, 2015 11:02 AM, David Pickett <d...@fugato.com> 
wrote:
   

 I dont expect them to ever sound as good as an Ambisonic recording, 
but I bought some SQ-encoded LPs today.  I get pleasant results 
playing them out of phase with the same on two rear channels at -6 dB.

My reason for writing is to ask whether anyone here knows what an SQ 
decoder actually did.  Despite all the BS Ben Bauer spouted when he 
presented it to the AES in London (or was it the BKSTS?), I seem to 
recall that it wasnt too sophisticated and perhaps, knowing this, one 
can synthesize something better than the above in a DAW.

Thanks in advant for any pointers.

David

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

[Eric Benjamin]

All of my spatial sound research has been self funded. As such, Im very 
sensitive to the budgetary issues. I had initially selected the then-current 
genelec 1028a for my array, but then I discovered that the JBL lsr25 had 
flatter response, greater low frequency extension, more acoustic output, and 
cost half as much. The pricing was such that JBL was far cheaper here than in 
Europe.

So, based on that, have a look at the LSR308. Its $250 instead of $350 for 
the genelec, and has an 8 woofer instead of a 3 woofer. Give it a 
listen!

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

[Eric Benjamin]

By all means. We'd like to hear them. Why not put them on Ambisonia?
http://www.ambisonia.com


--
On Sun, May 17, 2015 10:47 PM PDT Bo-Erik Sandholm wrote:

I have a number of  recordings using a tetramic and Tascam DR-680, the 
recordings are mostly:
church choirs among other recordings a full church with only singers  making a 
700+ choir singing Händel,
Jazz recordings ( Benny Goodman style)  in a club with audience eating and 
talking,
One Gospel recording with all of the public joining in,
A few short recordings of  rainy afternoons with a bit of thunderstorms from 
the Swedish archipelago.
A long recording of Early morning birdsong at sunrise 03:30, with absolutely 
no wind or man made background sounds, from a wooden glade in the archipelago.

Most of them can be distributed within this group and the ambisonic community 
if someone is interested and wants to listen.

Best Regards
Bo-Erik Sandholm



-Original Message-
From: Sursound [mailto:sursound-boun...@music.vt.edu] On Behalf Of Curtis 
Alcock
Sent: den 14 maj 2015 12:33
To: Surround Sound discussion group
Subject: Re: [Sursound] Converting 16 mic array recording to B format

Thank you everyone who contributed to answering my question. I am now fully 
convinced that using the DEMAND library would be next to useless and not worth 
the work involved in trying to make it (partially) usable. Pity really, as a 
database of everyday noise would be a useful resource - but only if it's done 
properly in the first place.

I am in the process of doing my own field recordings using a Brahma mic (been 
impressed with results so far: birdsong; a very noisy, reverberant restaurant; 
organ playing in a church) but as it will take some time to build up a 
complete library I was looking for some other material I could use, 
particularly every day environments.

Thanks, Richard, for your suggestion of ambisonia.com. I started there but was 
having trouble downloading the torrent files. (The internet access I have here 
(in a shared building) blocks torrent files so I need to download somewhere 
else then transfer.) Will continue to pursue this route.

Thanks again, everyone.

On 14 May 2015, at 19:11, Richard Lee rica...@justnet.com.au wrote:

 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] 3D Sound Labs Neoh headphones

[Eric Benjamin]

NEOH seems to be quite similar to the Rondo from Dysonics:http://dysonics.com/

The Rondo is a small device that can be attached to any headphone and provides 
head tracking. The rest of the product is a software player (RAPPR) that 
applies the head tracking info and communicates to Rondo via Bluetooth. So far 
as I know, its MacOS only.
Dysonics also has a microphone array http://dysonics.com/our-technology/

I haven't yet heard either of these products, but I intend to! 



 On Wednesday, April 22, 2015 10:13 AM, mgra...@mstvp.com 
mgra...@mstvp.com wrote:
   

 - Original Message -  Subject: Re: [Sursound] 3D Sound Labs 
Neoh headphones
From: Stefan Schreiber st...@mail.telepac.pt
Date: 4/22/15 12:05 pm
To: Surround Sound discussion group sursound@music.vt.edu

Stefan Schreiber wrote:

Ok, they could be a bit clearer. They could refer to anything specific 
 above 5.1/7.1, what they avoided. They could maybe have mentioned 
 Ambisonics, but most people never heard about.

 Therefore 3D audio formats and immersive.

 They could connect the headphones to a (binaural...) Mpeg-H 3DA decoder, 
 but same story here: The potential customers probably never have heard 
 of Mpeg 3DA. The music industry or what remains doesn't know a lot if 
 anything, etc.

 In fact: 3D Sound Labs should license (or obtain) a few real 3D audio 
 recordings, for demonstrational purposes. (We are getting into marketing 
 related questions.)

 Best,

 Stefan

 P.S.: Which gives some urgency to the question how to improve Ambisonics 
 decoders, and especially binaural Ambisonics decoders. You know that I 
 have said this again and again. Don't want to complain too much in 
 public, even if... ;-)

 P.S. 2: You have been living in a flat dream-world, Neoh... :-D



 delurksWhat really irks me are the binaural conference services, like 
BT+Dolby Voice or Voxeet. They pitch their service as being 3D audio, but they 
lack any concept of the vertical dimension. When I call them on that matter I 
get accused of being too fussy.
 
In reality, the 3D aspect of their marketing is really just something sweet 
to attract large enterprise or VC flies. /delurks
 
Michael Graves
 mgra...@mstvp.com
http://www.mgraves.org
o(713) 861-4005
 c(713) 201-1262
 sip:mgra...@mjg.onsip.com
 skype mjgraves
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Re: [Sursound] ENVELOP - 3D Sound, on Kickstarter.com

[Eric Benjamin]

Martin

There is a bit more information about the project at the envelop web site:

http://envelop.us/

Several of the proponents are members of Surround. Perhaps one of them would 
like to elaborate.

Eric

--
On Fri, Apr 10, 2015 9:46 PM PDT Martin Leese wrote:

Hi All,

I fell across this Kickstarter campaign.  I have
no connection, blah, blah, blah.  Does anybody
know more?  Below are some extracts.

Regards,
Martin
-- 
Martin J Leese
E-mail: martin.leese  stanfordalumni.org
Web: http://members.tripod.com/martin_leese/


ENVELOP - 3D Sound

https://www.kickstarter.com/projects/envelop/envelop-3d-sound

$27,333 goal

ENVELOP is a 3D sound platform that introduces an open source software
toolkit and educational initiatives to help any artist align their
creativity with new technologies for creating music in three
dimensions.

We are creating a next-generation 28.4 speaker audiovisual system that
features innovative technology to move sound in space around the
audience from any direction. ENVELOP is new environment for artists to
create and perform music in 3D surround sound.

The foundation of ENVELOP’s technology is Ambisonics

Not-for-profit and Open Source

Developers: Roddy Lindsay, Elan Rosenman, Christopher Willits, and Andrew 
Kimpel
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[Sursound] Recording B-format

[Eric Benjamin]

Ricky Graham ri...@rickygraham.net wrote: 


Does anyone have any experience recording B-Format with multiple microphones 
set to specific polar patterns (i.e. if you don’t have access to an ambisonic / 
tetramic?). Is it possible? If so, what are some of the issues / problems with 
this approach?

***
Along with Thomas Chen, I wrote two AES papers on this subject, The Native 
B-format Microphone parts I and II.  The abstracts can be seen here:
http://www.aes.org/e-lib/browse.cfm?elib=13348

http://www.aes.org/e-lib/browse.cfm?elib=13444


The gist of the papers is, as you say, how to record B-format if you don't have 
a soundfield microphone.  Aside from getting the microphones as close together 
as physically possible, it's important to do some post-processing to make the 
frequency responses of the various microphones more alike than they naturally 
are.  Most figure eight condenser microphones roll off in the bass at about 150 
Hz.  Most omni microphones make it down to 20 Hz or lower.  You can either 
boost the bass on the Fig 8s or roll-off the bass on the omnis. We used two 
systems.  One of them was comprised of lavaliere microphones (or hearing aid 
microphones) and the other was two Schoeps Mk8s and an Mk2.

One idea that we didn't try is that perhaps the omni (W) should be facing 
upwards.  Omnidirectional microphones, even very good ones like the Schoeps, 
are really not very omni.  So facing the microphone upwards guarantees that the 
response will be the same in every direction in the horizontal plane. The 
front-back (X) microphone can be placed above the omni, facing downwards.  That 
leaves the left-right (Y) microphone to be placed elsewhere.  There are no 
perfect solutions here!

If you would like copies of the two papers, just let me know.  I haven't looked 
at  them since we wrote them, but I don't think that we told any lies!

Eric Benjamin
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Re: [Sursound] Surroud sound with height demos

[Eric Benjamin]

Your request is intriguing. My question is: What sorts of demo material? I've 
made a number of such recordings with a soundfield microphone, none of which 
are in my possession at present. One of the recordings was of a male voice in a 
large studio room, with the subject ascending and descending a tall stepladder. 
I also made recordings of a fireworks bottle rocket being launched.  I also 
planned to make a recording from a footbridge overlooking a small stream.  A 
babbling brook from a height of 4 to 5 meters above.  Bird calls recorded in a 
canyon with the majority of the birds well above the microphone height.

Many such recordings give an impression of height, even when reproduced from a 
horizontal-only system.  This is presumably due both to expectation (where else 
would a plane be but above me?) and due to ground bounce.

Can you describe the type of recording that you would like?  I may be able to 
oblige you.

Eric


On Friday, January 9, 2015 7:00 AM, John Leonard j...@johnleonard.uk wrote:
 


Sascha,

I have a bunch of B-Format material with plenty of height information (mostly 
historic fighter planes) that I'm happy to provide for demo purposes. Is this 
of any use?

John

Please note new email address  direct line phone number
email: j...@johnleonard.uk
phone +44 (0)20 3286 5942


On 8 Jan 2015, at 10:38, Sascha Spors sascha.sp...@gmail.com wrote:

 Dear Sursounders,
 
 I am looking for publicy available demos of surround sound with height. I
 would like to demonstrate my students new developments in surround sound.
 The current trend towards reproduction with height is definetly very
 interesting in this context.
 
 However, I was not able to find demos on the net. Does anybody have an idea
 where to get demo material from? The particular loudspeaker setup used in
 the demo is not so critical since we have a rig with flexible mountings.
 
 greetings,
 Sascha
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Re: [Sursound] Ambisonic Cube reverberation attenuation with foam

[Eric Benjamin]

How have you determined that reflections from the room walls are the source of 
the perceptual problems?

To test for that, it might be best to try a temporary fix first.  I have 
successfully used moving blankets to prototype room treatment.  If moving 
blankets are installed away from the actual walls then they are extremely 
efficient absorbers, even down to low frequencies.  This can be done using 
hooks and twine and the amount of absorption can make the room almost anechoic. 
Perhaps they can be attached to the outside of your rig.

Moving blankets can typically be purchased for $10 each or about $80 for a 
dozen.


On Tuesday, October 21, 2014 1:52 AM, Bo-Erik Sandholm 
bo-erik.sandh...@ericsson.com wrote:
 


Covering the corners  - floor and walls + roof and walls with something 
diffusing or absorbing is probably the most important, a corner is a perfect 
for reflecting incoming sound back in source direction.
After the corners come the other 90 degrees angles bit hey are not as critical, 
but a soft longhaired carpet along the walls or on the walls up to around a 
meter height is good.

Best Regards Bo-Erik

-Original Message-
From: Sursound [mailto:sursound-boun...@music.vt.edu] On Behalf Of Tommaso 
Perego
Sent: den 21 oktober 2014 07:10
To: sursound@music.vt.edu
Subject: [Sursound] Ambisonic Cube reverberation attenuation with foam

Hello Everyone
I was wondering if you could please help me solve the following problem.

I have encountered reverberation issues with an Ambisonic installation, of 
dimension 5x5x2.5 meters (a squashed cube, so to speak).

I have noticed that reverberation is due to the proximity to the surrounding 
walls (7x12x6), causing imperfect appreciation of the spatial sound designs 
when heard in the middle of the cube.
Assuming that this is the correct understanding of the problem I was wondering 
if:

- surrounding  the cube with the following foam material

http://www.anyfoam.co.uk/sheet-foam.php 
http://www.anyfoam.co.uk/sheet-foam.php  (the acoustic foam)

would significantly reduce reverberation effect to better the definition inside 
the cube?

- where exactly would be best to put the foam? Would just the sides (excluding 
floor and ceiling) of the cube be enough ?

- should reducing the overall sound power improve the situation?


Looking forward to hear your opinion, I would greatly appreciate your help 
Thank you

kind Regards

Tommaso
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Re: [Sursound] Noise reduction on Ambisonic files

[Eric Benjamin]

Garth,

I wonder why it is that your recordings are so afflicted by noise.  The self 
noise spec for the SPS200 is 12 dBA, which is similar to that of other 
soundfield microphones from Soundfield.  While 12 dBA isn't noise free, it 
should be pretty quiet.  As a reference, the average threshold of detectability 
for microphone noise is about 6 dBA, assuming a natural recording scenario.  
That is, assuming that the sounds are replayed at the same level at which they 
occurred in the recording environment.

Of course, it may be that the microphone doesn't meet specifications.

I'm a bit confused by the recordings that you placed at
http://listen.ame.asu.edu/sonic_events.php


The first recording is labeled as no audio.  The second recording is labeled 
as you can hear Garth open his canteen and move some things around.  There's 
certainly a lot more noise in that second recording.  About 46 dB more, 
unweighted.  It would be interesting to try to perform some more controlled 
recordings to find out whether the noise is coming from the mic, or not, and 
whether it meets specifications.

Do you ever get to the SF bay area?

Eric Benjamin


On Wednesday, August 6, 2014 3:12 PM, Sampo Syreeni de...@iki.fi wrote:
 


On 2014-08-06, Joseph Anderson wrote:

 I take the noise profile from each individual A-format channel...

At the risk of sounding trite, what is noise? I'd argue that it isn't 
one thing, and that it's pretty difficult to define with mathematical 
precision. If you're talking about environmental background, then 
approaches like gating A-format or some other suitable directional 
representation of sound is a good idea.

If you're talking about tape noise instead, that isn't directional at 
all, at least until you get into directional masking calculations over 
what you can throw away without getting caught. In that case you'd want 
to operationalise what you consider noise, then find out an optimal way 
of extending that idea to B-format, and do the kind of joint processing 
Eero suggests.

The easiest way probably is to go with just W in the sidechain and equal 
gating for all the channels in the main one. The next step would be to 
do the same per frequency, and so on. However, in the ambisonic world, 
you'll then bump into a third source: the mic. Since the Soundfield 
works on differencing principles, W has a totally different noise 
profile from XYZ, and typically it only gets worse from there as the 
order goes up. (Or it doesn't; that depends wholly on the mic geometry.)

The point is, I don't think there is a monolithic thing called noise 
which can be just blindly reduced. There never was even in monophonic 
recordings, and the free degrees of freedom in your signal chain just 
multiply when you go through stereo to ambisonic. So, you need to be 
careful about which source(s) of unwanted hiss, distortion or bogus 
sources you're talking about, you'll have to develop computationally 
tractable models of both your utility signal and the noise, and only 
then can you really start to combine all of the machinery into something 
which actually works/sounds good.

E.g. when you expand/limit A-format, implicitly your noise model is a 
hiss which is directional to first order and your model of the utility 
signal is something like a strong, wideband directional signal near it, 
which makes directional sine-to-noise masking statistics relevant. Break 
those conditions and bad things will most likely happen.

So, try your approach on a two sine test signal, separated in frequency 
more than a critical band's worth. Pan one of the sines due front, and 
revolve the other one around at about 1Hz and say -6dB. Then add pink 
noise at about -10dB to each of the B-format channels independently. I'm 
rather sure that while your approach will work beautifully for the front 
signal alone when adjusted right, it'll lead to nasty, anisotropic noise 
pumping with the dynamic signal in place.

(Oh, and by the way, which A-format? As long as you're dealing with a 
perfect mic and linear, time-invariant filtering operation, you don't 
have to think about that because you can go willy nilly between A and B. 
But once you start applying this kind of processing, every possible 
orientation of the mic gives rise to a separate A-format. Which one 
should it be? The above example presumes one of the capsules is facing 
towards the reference. It gets much worse if you place the source 
directly between three adjacent capsules, in angle space.)
-- 
Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front
+358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2

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Re: [Sursound] Inexpensive USB multichannel sound card

[Eric Benjamin]

I can recommend the very inexpensive Sabrent SND-8, which is $19 at Amazon.  
It's not great, but it's certainly adequate!



 From: Alessandro Fogar sfo...@gmail.com
To: sursound@music.vt.edu 
Sent: Sunday, March 23, 2014 2:49 AM
Subject: [Sursound] Inexpensive USB multichannel sound card
 

Hi all,

can you please suggest me an inexpensive USB multichannel sound card (8ch)
to use in an installation ?

If possible with drivers for Linux, Win, Mac.

Many thanks in advance

Best

-- 
Alessandro Fogar

http://www.fogar.it
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Re: [Sursound] 'Quasi-flat' binaural

[Eric Benjamin]

David,

Intriguing, but I don't know exactly what it is that I'm supposed to be 
hearing!  A little descriptive text would be helpful.

Eric Benjamin



 From: dw d...@dwareing.plus.com
To: Surround Sound discussion group sursound@music.vt.edu 
Sent: Friday, January 10, 2014 6:23 AM
Subject: [Sursound] 'Quasi-flat' binaural
 

https://www.dropbox.com/s/iga71wluxfcb4i6/Untitled2.mp3
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Re: [Sursound] Encoding a 7.1 audio DVD ?

[Eric Benjamin]

Yeah...  You can put them on the disc, but you can't get them back.

Likewise, I could envision putting up to the full 64 channels of audio onto a 
disc by first encoding it as AAC, and then stuffing the AAC into a quasi-PCM 
file which would be entered into the linear PCM zone on the disc.  But again, 
how would you play it?

It seems to me that the easiest route for things like installations is some 
sort of miniature PC, like a Mac mini or an Intel NUC, and then an inexpensive 
8-channel USB audio device. I think that could be done for perhaps $300 to 
$400.  Beyond 8-channels it gets a little more expensive.



 From: KK Proffitt k...@jamsync.com
To: Surround Sound discussion group sursound@music.vt.edu 
Sent: Tuesday, December 17, 2013 12:14 PM
Subject: Re: [Sursound] Encoding a 7.1 audio DVD ?
 

While eight channel PCM was in the original DVD spec, it was, to my knowledge, 
never implemented in commercial players.

Best,

KK
-
KK Proffitt
President, JamSync, Nashville
Owner/Trustee: Hoodley Creek Farm, Afton, TN
k...@jamsync.com
www.jamsync.com
www.tnfilmlocations.com
www.surroundeffects.com
twitter: jamsync
twitter: kkproff
facebook: http://www.facebook.com/pages/JamSync/102314633504
phone: 615-320-5050






On Dec 16, 2013, at 10:20 AM, Augustine Leudar augustineleu...@gmail.com 
wrote:

 Seeing as there is a dearth of 8 channel players - I was thinking I could
 just use a 7.1 DVD on a loop for an eight channel sound installation - as
 long as I can send a seperate audio signal to each of the 8 (the LF sends a
 full range signal too) there shouldnt be a problem. I encoded 5.1 DVD ages
 ago  and I vaguely remember I needed several programs, one for encoding
 AC3, one for authoring etc etc - does anyone know  programs would I need to
 encode a 7.1 DVD ?
 best,
 Gus
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Re: [Sursound] Hector bird recording - SoundCloud

[Eric Benjamin]

 Is the W level correct? I am finding myself turning up my W knob 3-6dB 

I'm tempted to agree.  Of course it's difficult to be certain.



 From: dw d...@dwareing.plus.com
To: sursound@music.vt.edu 
Sent: Friday, November 22, 2013 6:21 AM
Subject: Re: [Sursound] Hector bird recording - SoundCloud
 

On 21/11/2013 19:08, Aaron Heller wrote:
 I took the liberty of merging them into 4-channel files and putting them on
 my server, which might be easier to access than the skydive (the UI was in
 Japanese for me, fortunately I recognized the character for 'down')

    http://ambisonics.dreamhosters.com/01-Birds_WXYZ-110425_0119.wav
    http://ambisonics.dreamhosters.com/05-Music_WXYZ-110425_0127.wav

 They sound quite nice.  In Harpex, you can clearly see the locations of the
 singers, percussion, and birds.  Impressive!

 Thanks...

 Aaron (hel...@ai.sri.com)
 Menlo Park, CA  US

Is the W level correct? I am finding myself turning up my W knob 3-6dB 
relative to other recordings before it sounds good..
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Re: [Sursound] New Ambisonic VST Plugins

[Eric Benjamin]

 David Pickett d...@fugato.com wrote:



 How does one record in third order (or indeed any order above first order)?  
 What kind of microphone array does one need, for instance, for 3rd order 
 with no height information (WXYUVPQ)? 
 Is there a native format method for HOA or is it all extended A format, 
 with conversion through matrices?


All excellent questions.  It is not quite as obvious how to record any order of 
Ambisonics above first order.  It will require some sort of microphone array 
and post-processing.  One of my favorites is the array described by Craven, 
Lawe and Travis in:
Microphone arrays using tangential velocity sensors
P.G. Craven, C. Travis, M.J. Law
We introduce a new class of 3D microphone arrays that use symmetrical 
arrangements of tangential velocity sensors.  Use of velocity sensors allows 
these arrays to recover spherical harmonics of a given degree with less 
low-frequency boost than when using pressure sensors only.  As an example we 
present a symmetrical array of twelve velocity sensors that resolves the eight 
harmonics of degrees 1 and 2.  A second-order spherical microphone can now be 
constructed by combining this array with one or more pressure sensors that 
provide the missing harmonic of degree 0.
http://ambisonics.iem.at/symposium2009/proceedings/ambisym09-craventravis-tangentialsphmic.pdf/at_download/file


The other practical method for constructing an array that produces higher order 
spherical harmonic outputs is to use a group of omnidirectional microphones on 
a sphere, such as the commercially available Eigenmike:
http://www.mhacoustics.com/products


There are other methods.  It's still early days for this technology.

Eric Benjamin
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Re: [Sursound] Ambisonic first approach

[Eric Benjamin]

I'm trying to get an overall sense of what people's objections are.  It seems 
that the mini XLRs are OK, but just OK, but suspending the phantom power 
eliminators has caused problems for users?  I've used mini XLRs in lots of 
projects for more than 25 years and found them to be OK.  There are lots of 
different manufacturers now, as opposed to back then, and thus some of them are 
not as good as others.  And mini XLRs as opposed to regular XLRs don't inspire 
confidence.  They tend not to make a really positive contact when snapped into 
the socket.

As for regular XLRs, there are no 8-pin XLRs.  At least I don't know of any.  
And 6-pin and 7-pin XLRs are very rare.  I can't get them in the type that I 
prefer.

What sort of multipin connectors do folks here like?

Eric Benjamin



 From: Michael Chapman s...@mchapman.com
To: Surround Sound discussion group sursound@music.vt.edu 
Sent: Sunday, November 17, 2013 4:19 AM
Subject: Re: [Sursound] Ambisonic first approach
 

 --On 16 November 2013 06:31 + Michael Chapman s...@mchapman.com
wrote:

 --fiddling to get the correct one of (?)24 possible permutations of the
four capsules to the snake is a !  Especially in dim lighting ...
and you didn't bring a lamp ... and ...

 A dab of four coloured paints on the plugs and sockets makes this so
much easier!

 That's why I was asking here about colour codes a few years ago (the
answer is make up your own!).

 Paul Hodges


To preserve the best traditions of ambisonics ... only if they clash with
someone else's.

More seriously ... my cones give out before my rods ... so big black and
white numbers (or any glyphs) is still easier.
Remembering the torch is even easier.
And ... of course ... a six, seven, eight pin XLR 'plug and play' would
cut out the 'human' element altogether ...

Michael



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Re: [Sursound] Eigenmike (Fons Adriaensen)

[Eric Benjamin]

The eigenmike is supplied with a software application called EigenStudio, which 
allows the user to output various beams with selectable directivity.  Those 
outputs aren't, or mostly aren't spherical harmonics.  Instead, one has 
available cardioid, hypercardioid and supercardioid outputs up to 3rd order.  
Of course it is possible to derive the spherical harmonics from those.

Eric Benjamin


- Original Message -
From: Stefan Schreiber st...@mail.telepac.pt
To: Surround Sound discussion group sursound@music.vt.edu
Cc: 
Sent: Monday, June 17, 2013 8:52 AM
Subject: Re: [Sursound] Eigenmike (Fons Adriaensen)

For clarification:

Could the eigenmike also do some 4th order recording (in a real sense, 
not giving some 4th order output), or is it a 3rd order microphone?

Thanks,

Stefan



Fons Adriaensen wrote:

On Sat, Jun 15, 2013 at 09:21:55PM +0100, Paul Power wrote:

  

so it seems that the microphone is still not a perfect solution for
HOA recording and requires some highly technical tinkering before it
will give half decent results.
    


It depends very much on the application. Some of those recordings
made by Farina's team  are quite good, for example the collection
of 'ambient sounds' of the city they made earlier this year. But
the musical ones so far failed to convince me. It's the processing
SW that isn't yet what it could be - I've no doubts about the
quality of the mic itself.

I've been present at some of the music recordings, so I can compare
my own impression of the event and the reproduction. One of these 
was at the opera house in Parma. The mic was placed just behind the 
conductor, and maybe half a meter above his head. Not and ideal
position, but it provides a reference point. I was sitting in the
first row, less than a meter from the mic stand. The recording was
'decoded' using a number of spot mics looking down on the orchestra
and four or so covering the stage (an angle of 120 degrees or so as
seem from the mic). The spot mics were then panned into 3rd order AMB,
and also used as primary sources in the WFS system of the Sala Bianca.

In particular the stage mics needed quite some EQ in order to sound
more or less natural, but even then the result lacked the presence
of the original sound. Parma's opera house is very 'dry' - the result
of the layers of plaster added with each restoration without removing
the older ones. So sitting in the first row you really get a sound
that is quite 'direct', I wouldn't want a seat there normally. But it
seemed impossible to reproduce anything like that.

Ciao,

  


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Re: [Sursound] Nevaton microphones

[Eric Benjamin]

 did you ever look through what Z really does to you *encoding* equations?
 because they're always a bit spread out even vertically, not all of that 
problem can be remedied.
AND
 The only way to really get that distance calculation right is to employ 
periphony

You are undoubtedly correct.  But I look at this problem more from a practical 
point of view.  A horizontal-only presentation of a full 3D space is just 
wrong. 
 But practically speaking, only occasionally do we have the opportunity to 
present a recording or a composition periphonically.  So we must accept the 
compromise.  The distance-related errors I don't see as being such a big deal 
because, especially at low order, we have to accept the summation errors which 
lead to a small sweet spot.  But how do we perceive the space, especially when 
there were sources in the original recording that were out of the horizontal 
plane?  Or, in my own recording work, when there is reverberation coming from 
out of the horizontal plane?  This leads to a concentration of energy right at 
the horizon.  In recent work, Aaron Heller and I have been trying to produce 
decoders that work 'better' than traditional decoders for arrays that only 
partially cover periphonic space, arrays like a hemisphere or a tilted 
hemisphere in a concert hall with banked seating.  If the original sound source 
started out above the horizon but then moves downward to eventually go below 
the 
horizon, we can reproduce the direction and timbre of the sound pretty well in 
these arrays when the source is above.  But when it dips below the horizon we 
have to make choices as to which wrong behavior we will allow.  We could enact 
a strict 'no-fly' zone; sources just disappear when they dip below the horizon. 
 Or they can 'stick' at the horizon, which tends to be the natural behavior of 
Ambisonic decoders.  Or you can have them gradually fade away as they go lower. 

This is also a problem with some of the periphonic arrays that I have tried in 
the past.  I still like the 30 degree tri-rectangle, because it fits in my 
listening room. But it has the fault that sources tend to stick at +30 degrees 
and at -30 degrees.  In practice, this doesn't seem to be so much of a problem 
with naturally recorded program material even though there must be a 
concentration of reverberant energy at those two elevations.

AND
 advance a bit towards an understanding of higher order compatibility formats, 
and in the process, of how to optimally and scalably encode ambisonics?

I think that we still don't understand well enough what the perceptual effects 
are of the compromises we make in decoder and system design.  There probably is 
a good deal of progress still to be made in this area.

Eric Benjamin


- Original Message 
From: Sampo Syreeni de...@iki.fi
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Sat, June 1, 2013 6:55:04 AM
Subject: Re: [Sursound] Nevaton microphones

On 2013-05-31, Daniel Courville wrote:

 I always insist on recording 'Z', and then almost never end up using it...
 
 Not even to look down or up in a stereo decode? I use the Z quite often (if 
not always) when recording large ensemble and the SF mic is more than 10 feet 
off the floor.

BTW, did you ever look through what Z really does to you *encoding* equations? 
Formally, in order to arrive at proper pantophony you always have to either 
reject Z fully or purposely subtract it from the whole B-format signal set. 
Otherwise, even assuming perfect coincidence, your W will have directionally 
aliased components from the above and the below. For the most part that isn't 
noticeable in e.g. concert work where you have a wide and loud array of early 
arrivals right in the horizontal layer. But the theoretical error easily bite 
even with a simple walkaround where not everything is in the horizontal plane, 
including close vertical room modes.

That particular problem has then also been used to attac the ambisonic system 
as 
a whole. I believe I told about Christof Faller's analysis of why ambisonic 
can't work, a few years back, didn't I? Which I followed live at what is now 
Aalto University, and then Teknillinen korkeakoulu (lit. technical high 
school, formally Helsinki Polytechnic).

How Christof saw it was much from the WTF point of view. There, if you 
reproduce 
a point source in the horizontal plane only using a horizontal array of 
speakers, you will get the angle of arrival right, but the normal attenuation 
suddenly acquires an extra 3dB/per normalized distance factor. In WFS they 
purposely compensate for that with their linear and rectangular arrays. But 
very 
few analyses really go into where that factor comes from, or how it could be 
avoided, or what it's really about. The pantophonic analysis of ambisonic 
doesn't go there either, even if it really, *really* should.

The basic problem is that you just can't in 3D space radiate a 3D soundfield 
which fails to collimate in at least

Re: [Sursound] Nevaton microphones

[Eric Benjamin]

Two dual-diaphragm capsules mounted one above the other.  By appropriate 
summation one can in principle achieve any first-order pattern pointed in any 
horizontal direction.  Thus it's possible to output the horizontal part of 
first-order B-format.  Dual-diaphragm capsules can have very good 
figure-of-eight patterns, even at the large size of these capsules.  But the 
omni component tends to be quite impaired.  But if omni is made by summing 
together the outputs of all four diaphragms then it should be somewhat 
improved. 
 This could work fairly well.

I haven't used this microphone but I've done similar things, with good success.

The lack of a vertical component is an interesting conundrum.  I always insist 
on recording 'Z', and then almost never end up using it...

Eric Benjamin


- Original Message 
From: Emanuele lamacchiaco...@yahoo.it
To: sursound@music.vt.edu
Sent: Thu, May 30, 2013 8:11:31 AM
Subject: [Sursound] Nevaton microphones

Hello,

anyone tried these multichannel mic?

http://nevatonmics.com/mics_multichannel.php

I wonder what the quad configuration is.

Thanks.

Ema
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Re: [Sursound] [ot] hard to find papers

[Eric Benjamin]

They should be in your in-box by now.  I trust you were able to find the 
patents?


- Original Message 
From: Sampo Syreeni de...@iki.fi
To: sursound-list sursound@music.vt.edu
Sent: Tue, May 21, 2013 5:27:36 PM
Subject: [Sursound] [ot] hard to find papers

Does anybody happen to have the Dolby A and SR papers handy? They seem to be 
particularly difficult to find without access to the AES archives.
-- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front
+358-50-5756111, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2
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Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)

[Eric Benjamin]

On 30 Apr 2013, at 04:56, David Pickett wrote:

 A standalone Windows app that would decode Dolby-A encoded wavefiles and 
 output 
a restored non-Dolby 24-bit wavefile would be most useful.  I have several 
recordings that I have had transfer to hi-res files still in Dolby-A format.
 ... even if such a program were command line only and needed to be left 
overnight to cook!

Being a fan of doing things the easy way, I'd recommend just buying a Dolby 
Model 363 NR unit which does both A type and SR.  At any point in time there 
are 
typically a dozen or so available on Ebay for prices in the range of $150 to 
$300.  It's difficult to model something like Dolby NR in DSP because the 
algorithm is defined by a circuit.  You would need to very carefully benchmark 
a 
working decoder in any case because neither the patent or the JAES article 
really tell you how to do it.
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Re: [Sursound] what mics do you use?

[Eric Benjamin]

I'd like to expand just a bit on what Dave said.

The narrowing of the pattern of microphones at high frequencies is equivalent 
to 
the addition of higher order spherical harmonics into the directionality.  I 
recently went through the exercise of decomposing the pattern of a 1 capsule 
into its spherical harmonics and it took up to 10th order to do a good 
approximation at 16 kHz.  If one were to derive either an omni (monopole) or a 
Figure 8 (dipole) by adding or subtracting capsules then half the higher order 
harmonics would remain, resulting in a polar pattern that differs greatly from 
what was desired.  This is true even assuming that you could make the capsules 
coincident, which you can't.  A mental model of a soundfield microphone at HF 
is 
of four beams pointing out into space from the locations of each of the 
capsules.

The non-coincidence is of course a separate effect.  If we were to use perfect, 
point-sized capsules then they could conceivably have perfect cardioid 
patterns. 
 But the spacing effects are still there.  I've measured most of the available 
soundfield microphones to determine the value of r.  It's a little difficult 
because the center of the array isn't available, but one can measure from the 
center of one diaphragm to another and get r from that. If the capsules are 
cylinders of length l and diameter d, then

r = l +.2887d

SF MkIV and MkV1.47 cm (from literature)
SF SPS2002.71 cm (from measurement
AGM MR1 and MR22.27 cm (from measurement)
Tetramic1.77 cm (from measurement)

Note that r for the SPS200 is almost twice the value for the MkIV type design. 
 Long capsules make things worse!  I've built prototypes here with r = .7 cm, 
but none of those are ready for use.

Finally, Aaron Heller and I presented two papers at the 133rd AES convention 
that deal with some of these matters, in particular the diffuse-field response. 
 They are:

Calibration of Soundfield Microphones using the Diffuse-Field Response
http://www.aes.org/tmpFiles/elib/20130429/16453.pdf

A second-order soundfield microphone with improved polar pattern shape
http://www.aes.org/tmpFiles/elib/20130429/16470.pdf

I hope that the illustrations in these papers will make clearer what we've been 
talking about.  Either Aaron or I will be happy to send a copy to anyone who is 
interested.

Eric Benjamin


- Original Message 
From: Dave Malham dave.mal...@york.ac.uk
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Mon, April 29, 2013 9:28:32 AM
Subject: Re: [Sursound] what mics do you use?

Ok, you have two problems with large capsules. Firstly there's the standard
one of the basic directionality going off. The directional patterns of any
capsules degrades as the frequency goes up, due to interference effects,
and this happens at lower frequencies with larger capsules. Secondly if you
are deriving B format signals (or anything similar) from a capsule array,
the wider the separation the lower the frequency at which the derivation
fails which is why the tetramic produces such good patterns to such high
frequencies compared with the actual Soundfield. However, the larger
capsules of the Soundfield are a lot quieter and nicer' simply because
they are based on better quality and larger diaphragm  capsules - so, you
pays your money and makes your choice.

Dave

On 29 April 2013 15:56, Ronald C.F. Antony r...@cubiculum.com wrote:

 On 29 Apr 2013, at 02:33, Dave Malham dave.mal...@york.ac.uk wrote:

  but the 30+
  mm size will seriously mess with the high frequency response f any
 derived
  horizontal only B format.


 Could you please elaborate on the expected effects from the larger
 capsules?
 Trying to figure out if that would result in something one can live with,
 or something that turns it useless.
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-- 
As of 1st October 2012, I have retired from the University, so this
disclaimer is redundant


These are my own views and may or may not be shared by my employer

Dave Malham
Ex-Music Research Centre
Department of Music
The University of York
Heslington
York YO10 5DD
UK

'Ambisonics - Component Imaging for Audio'
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Re: [Sursound] what mics do you use?

[Eric Benjamin]

 On 23 April 2013 21:05, Matthew Palmer palme...@mymail.vcu.edu  wrote:
  Thanks for the heads-up. Which mics do you use?
 


My 2 cents worth.  For binaural recording I have used the Bruel and Kjaer type 
4101 setup:
http://www.bksv.com/products/transducers/acoustic/binaural-headsets/4101.aspx

It's a sort of stethoscope-like arrangement with two matched DPA 4060 capsules 
located at the ear canal entrances.  I mention the 4101 because it brings up a 
number of important points, even though it is financially out of range for most 
applications.  The capsules are very small, so they can be located at the ear 
canal entrance and yet not block the entrance of sound so the user can still 
hear.  BK provides calibration information which allows you to equalize the 
resultant recording to get the correct diffuse-field response.  About 30 years 
ago Gunther Theile, et al, worked to have a flat diffuse-field response be 
designated as the correct response for headphones.  
http://www.aes.org/e-lib/browse.cfm?elib=5233

But for binaural the key is to have the response measured at or in the ear 
canal 
be the same as what would have been experienced had the listener be present in 
the original recording environment.  

If we use miniature omni microphones with flat free-field response for binaural 
recordings then the spectral response at the ear canal entrance won't be flat, 
either if measured in the free-field or if measured in a diffuse field.  And 
even headphones which are specified to have flat diffuse-field response don't 
really have flat diffuse-field response.  But if we play back a particular 
binaural recording then what we would like to have happen is to have the 
spectrum measured with the binaural microphones at the user's ear, when playing 
back the binaural recording, be the same as what is on the recording.  So if 
the 
headphones have, say a rising response, we can EQ that out in order to achieve 
the same response as in the recording.

All of this makes a great deal of difference in what is heard.  Of course we'd 
like to be able to transmit that recording from person to person and have 
everyone hear the same thing.  But that won't happen because headphones differ 
so much.  Nonetheless, good binaural recordings tend to sound striking when 
played back over good headphones regardless of the fact that the headphones 
aren't all the same.  It's worth noting here that the concept that having a 
flat 
diffuse-field response for headphones is the best thing is under some debate at 
present.  A paper to be presented at the upcoming 134th AES convention should 
have some interesting new data:

http://www.aes.org/events/134/papers/?ID=3474

Listener Preferences for Different Headphone Target Response Curves—Sean Olive, 
Harman International - Northridge, CA, USA; Todd Welti, Harman International - 
Northridge, CA, USA; Elisabeth McMullin, Harman International - Northridge, CA 
USA
There is little consensus among headphone manufacturers on the preferred 
headphone target frequency response required to produce optimal sound quality 
for reproduction of stereo recordings. To explore this topic further we 
conducted two double-blind listening tests in which trained listeners rated 
their preferences for eight different headphone target frequency responses 
reproduced using two different models of headphones. The target curves included 
the diffuse-field and free-field curves in ISO 11904-2, a modified 
diffuse-field 
target recommended by Lorho, the unequalized headphone, and a new target 
response based on acoustical measurements of a calibrated loudspeaker system in 
a listening room. For both headphones the new target based on an in-room 
loudspeaker response was the most preferred target response curve. 
Convention Paper 8867 

More about ambisonic recording in a subsequent email.

Eric Benjamin
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Re: [Sursound] what mics do you use?

[Eric Benjamin]

 On 23 April 2013 21:05, Matthew Palmer palme...@mymail.vcu.edu  wrote:
  Thanks for the heads-up. Which mics do you use?
 

Another 2 cents worth.  For Ambisonic recording there are a few alternatives. 
 Soundfield research has several models, mostly based on the original Calrec 
design.  
http://www.soundfield.com/products/mkv.php
http://www.soundfield.com/products/sps422b.php
http://www.soundfield.com/products/dsf2.php

There is another model, the SPS200, which uses MBHO capsules.

http://www.soundfield.com/products/sps200.php

From the competition we have:
http://www.core-sound.com/TetraMic/1.php

Also, there was a very expensive tetrahedral microphone array manufactured by 
AGM, called the MR-1:
http://www.agmdigital.com/page42/page22/page22.html

which is a tetrahedral array constructed of DPA 4012 type capsules.  I mention 
this, even though it is no longer offered by AGM, because DPA has told me that 
they have stock of the tetrahedral hardware, and if I want another one I can 
simply send them a check.  I didn't get so far as to ask how big the check 
should be...

As it happens, I've used all of these.  The AGM MR-1 has a little trouble due 
to 
the size of the array, but at lower frequencies its performance is flawless.

And there are a number of home-built alternatives.  An probably some others.

At which point I should point out that the quality of the results depends most 
critically on the calibration of the system, and secondarily on the size of the 
array.  As discussed in previous emails of the past couple of weeks the arrays 
very well up to a critical frequency 
f = c/2pi*r , where r is the radius of the array.  What this means is that a 
tetrahedral array the size of the original Calrec design works almost perfectly 
(if properly calibrated) up to about 7 .3 kHz, pretty well up to 10 kHz, and 
then goes to hell.  If this sounds bad, then just look at the polar patterns of 
a typical omni microphone.  A half-inch omni gets bad above 7 kHz or so.  A 1 
Large-diaphragm microphone is pretty horrible above about 4 kHz except in its 
figure-eight patterns.

To reiterate, one of the primary things that the mfgrs of tetrahedral 
microphone 
arrays have to offer is the accuracy of their calibration.  The inescapable 
variations in the frequency response which result from the size of the array, 
not the quality of the capsules, require that the array be calibrated 
(equalized) to get proper response.

Eric Benjamin
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Re: [Sursound] Surround formats and lossy compression

[Eric Benjamin]

 Has anyone experienced odd artifacts while doing hybrid mixing and where 
 sound 
files stored in lossy formats were converted to wav files?
No, but I only use lossy compression when I have no other choice.  Using 
lossless compression removes one possible source of error.

 Are there file formats that should be avoided as far as psychoacoustic 
 research 
goes?
See answer to question below.

 Are all lossless formats more-or-less equal in terms of 'purity'?
Did you mean to say lossy, or lossless?

Lossless compression is without error, assuming that basic assumptions of the 
coder are not violated.  Some of the commercial processes have modes that are 
'mostly' lossless.  I a streaming application, such as on a blue-ray disc, 
there 
is some maximum cap on bitrate.  If the audio source is highly random, that is 
to say that it's mostly noise, then the compressed data rate can be too large 
for the medium and then the codec is forced to enter into a lossy mode.  This 
is 
highly unlikely to happen except for some high sample-rate cases where most of 
the bandwidth above 20 kHz is comprised of noise.

Lossy compression is, of course, designed to make a representation of the 
original audio at some lower data rate.  Depending on the codec and on the data 
rate the decoded copy of the audio has errors in it, errors which are intended 
to be inaudible.  If the data rate is lower, then there is more error and the 
chances that it will be audible at any particular point in the recording are 
increased.  You can fool most of the people most of the time.  As the data rate 
is increased, the percentage of people who can't hear the artifacts approaches 
100%.  Also, the quality of any particular codec will depend on 
the implementation of it.  Codecs are specified primarily by their decoders. 
 That is, the decoder must be able to decode data streams with the specified 
syntax and tools.  It has typically been the case that after the introduction 
of 
a new codec that the encoders continue to get better, so that the overall 
performance at a given data rate is better using some later 
encoder implementation than it was originally.

All of this has been the subject of contentious debate for two decades.  There 
are plenty of people with opinions but relatively few of them who have taken 
part in any controlled listening tests.  Many individuals seem to have listened 
to a few bad examples and then made a long-term decision as to what works and 
what doesn't.   As an example, MP3 at 96 kbps is likely to have audible 
artifacts depending on what is encoded, what the reproduction system is like, 
and who is listening.  MP3 at 192 kbps is very unlikely to have audible 
artifacts, although certain tricky program material can cause audible artifacts.

It thus becomes the case that one might wish to rate codecs in terms of bit 
rate 
necessary to achieve a given quality level.  In that case a ranking might be 
AAC 
 DD  MP3  DTS.  All of them can achieve good quality but some of them 
 require 
a higher data rage to do it.

If the audio is processed after the low bit rate compression some of the 
assumptions about masking made by the designers of the codec may be violated. 
 This may result in artifacts being more audible than otherwise.  Have a listen 
to the AES demo disc.

http://www.aes.org/publications/technical/DigitalAudio.cfm

I see that this disc is a different one than what I worked on several years 
ago, 
but it looks as though it should have some good stuff on it.

One final comment.  When I watch digital broadcasts, say the ones over DirecTV, 
I see continuous video artifacts yet I never hear audio artifacts.  In 
situations with recorded or broadcast media one doesn't have knowledge of what 
the original program looked like or sounded like.  But if I hear a sound that 
doesn't normally occur in audio, or if I see an artifact like macro-blocking 
or halos or the screen door effect, I know that wasn't in the source originally.

Eric Benjamin (not to be confused with Eric Carmichael!)


- Original Message 
From: Eric Carmichel e...@elcaudio.com
To: Sursound sursound@music.vt.edu
Sent: Fri, April 5, 2013 5:52:36 PM
Subject: [Sursound] Surround formats and lossy compression

Greetings to All:

When it comes to surround sound coding/decoding, I never make a peep because 
I'm 
ignorant on the topic. However, a friend who heads the Dept. of Audiology at a 
children's hospital had asked a question regarding MP3s. Although the MP3 
format 
may be nothing more than a distant relative to surround formats, the thought of 
using lossy file types in research studies utilizing surround-sound stimuli 
does concern me. I answered my friend's question (re MP3s) as best I could, and 
the answer is shown below (I copied and pasted it verbatim--sorry for it's long 
length). Some of the concerns outlined below may or may not apply to surround 
sound (?).

Has anyone experienced odd artifacts while doing hybrid mixing

Re: [Sursound] Looking for a very, very old thread - mics pointing inwards?

[Eric Benjamin]

I'm still confused as to what is being described.  Could it possibly be this?

http://ambisonics.iem.at/symposium2009/proceedings/ambisym09-pollowmasiero-musicalinstrumentdirectivity.pdf/at_download/file


or something similar?

Eric Benjamin 



- Original Message 
From: Eero Aro eero@dlc.fi
To: sursound@music.vt.edu
Sent: Mon, March 4, 2013 7:08:31 AM
Subject: Re: [Sursound] Looking for a very, very old thread - mics pointing 
inwards?

 I'd be interested if you do turn up a reference.

The setup is used by some people I know. It works, if used carefully and
properly. Sorry, no references to point to.

 maybe it was in Resolution
 magazine.  I thought I'd kept the article, but I can/t find it in my file,
 and there doesn't seem to be anything on the web.  Maybe that's what you're
 thinking of?

There have been some articles in Sound On Sound about somewhat similar
subjects. However, these are not what I am looking for.
http://www.soundonsound.com/sos/aug01/articles/recacgtr0801.asp
http://www.soundonsound.com/sos/apr10/articles/acguitar.htm

Thanks for the comments.

Eero
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Re: [Sursound] Unusual Binaural Head

[Eric Benjamin]

This seems to me to be a bit like a device created by the folks at CIPIC, at 
the 
University of California at Davis:

http://www.news.ucdavis.edu/search/news_detail.lasso?id=7058

They didn't use multiple pinnae, but they did have multiple 'ears' around the 
circumference of a cylinder.  There were several AES publications on this about 
9 years ago.

Also the subject of a US patent:
http://www.google.com/patents/US20080056517?dq=V+Ralph+Algaziei=OhYaUdfkBaOpiQKI2oHoCg


Eric


- Original Message 
From: michael noble loop...@gmail.com
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Tue, February 12, 2013 12:02:51 AM
Subject: [Sursound] Unusual Binaural Head

I've come across mentions of binaural heads before but this one looks like
something different. Anybody have an idea what is going on here:

http://now.lincoln.com/2013/02/sound-technology-gets-a-human-touch/
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Re: [Sursound] Which order (but not extactly high order)?

[Eric Benjamin]

More opinion to add into the mix.

There are really tiny microphones which are also directional.  An example of 
the 
case in point is the Knowles FG which is available with several different 
directivities.  
http://www.digikey.com/Web%20Export/Supplier%20Content/Knowles_423/PDF/knowles-an-dfg-microphone.pdf?redirected=1


The DFG is about 2.6x2.6 mm in size, which is small enough to lose under your 
thumbnail!  It's available with 3 different directivities, one of which is 
approximately a cardioid.  The directivity is maintained pretty well over the 
audio bandwidth.  Does the directivity change when it's located next to an 
object?  Yes, and that's one of the main challenges in designing a microphone 
into a hearing aid.  The self noise is fairly low, at least for the omni 
version 
which has a self noise of about 25 dBA.  The trouble with using these for 
ordinary audio work is that the bass rolls off below about 7 kHz.  That 
necessitates a substantial amount of equalization to restore flat free-field 
response.  That has a negative effect on the SNR.  It's also worth noting that 
these hearing aid microphones typically can only go to about 105 dB SPL or so 
when configured as directed by the manufacturer.  Smart circuit design can 
alleviate that problem.

Hearing aid microphones illustrate a certain sweet spot in microphone design. 
 Comparing the Knowles FG omni to a modern small diaphragm recording microphone 
such as the KM83 shows that it has about 12 dB more noise (25 dBA vs. 15 dBA), 
a 
bit more low-frequency roll-off (the ones that I've tested are -3 dB at 10 Hz), 
less high frequency bump (+5 dB at 12 kHz vs. +8 dB at 10 kHz) and a reduced 
maximum input level (105 dB SPL vs. 140 dB SPL).

The main point is that hearing aid microphones are small enough to retain their 
directional characteristics all the way up to the top of the audio bandwidth, 
and also small enough to be put in places where other microphones just won't 
fit.  Whether they are good enough for a particular recording or measurement 
purpose depends on the exact needs of that purpose.

Eric



- Original Message 
From: Fons Adriaensen f...@linuxaudio.org
To: sursound@music.vt.edu
Sent: Wed, November 7, 2012 2:49:58 PM
Subject: Re: [Sursound] Which order (but not extactly high order)?

On Mon, Nov 05, 2012 at 03:59:19PM -0800, Eric Carmichel wrote:

 I would like to model microphone pickup patterns in conjunction
 with HRTFs and Ambisonic recordings that I've made. To give a
 specific example, I would like model a miniature supercardiod
 mic, pointed forward, that is located proximal (or superior)
 to the pinna.

A tricky problem, and I don't have an immediate answer to it.
But I do have question. Are these miniature mics *really*
directional, or do they become directional by being deployed
in the way they were designed to be, e.g as you describe ? 

The reason I ask is because a) making a really small mic 
directional while preserving other required performance
characteristics (e.g. sensitivity or self-noise) is not
an easy thing given the physics, and b) when a mic is
placed very close to any object having the size of a 
human head its polar pattern is likely to be modified by
the presence of that object, at least at medium and high
frequencies.

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)

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Re: [Sursound] higher order ambisonics over 8 to 10 loudspeakers

[Eric Benjamin]

Fons Adriaensen wrote:

 for anything based on energy vectors the angle between the speakers can't be 
too big. 

Is there a good reference for that important point?

Eric


- Original Message 
From: Fons Adriaensen f...@linuxaudio.org
To: sursound@music.vt.edu
Sent: Mon, July 9, 2012 6:48:40 AM
Subject: Re: [Sursound] higher order ambisonics over 8 to 10 loudspeakers

On Mon, Jul 09, 2012 at 11:13:04PM +1000, GP wrote:

 If the min is (N+1)².  
 Surely for 3rd order that is 
 (3+1)² = 16 speakers?

The minimum is (M + 1)^2 for 3D, and (2 * M) + 1 for 2D, but

- You better use at least on more,
- For 3D, the minimum is 8, even for first order. That is because
  the the equations above assume a systematic decoder, but a decoder
  should be systematic only at LF, and for anything based on energy
  vectors the angle between the speakers can't be too big. 
  
Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)

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Re: [Sursound] List of open source software for wavefield synthesis

[Eric Benjamin]

I'd be very interested in hearing about your hardware.  That many DAC channels 
and loudspeakers is challenging!


- Original Message 
From: Augustine Leudar augustineleu...@gmail.com
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Thu, July 5, 2012 7:59:21 AM
Subject: [Sursound] List of open source software for wavefield synthesis

Hello all,
I am currently making a WFS setup and would like to check out what software
is available (windows,mac or linux) . Could we compile a list of software
(preferably open source) avialable for wavefield synthesis on this thread,
thanks ,
Gus
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Re: [Sursound] preferred (small) linux distro for audio?

[Eric Benjamin]

I have almost no experience in this, but it seems like this discussion would be 
incomplete without mentioning the Planet CCRMA distribution:

http://ccrma.stanford.edu/planetccrma/software/ 

written with too much blood in my caffeine stream.

Eric


- Original Message 
From: Dave Malham dave.mal...@york.ac.uk
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Wed, July 4, 2012 3:42:11 AM
Subject: [Sursound] preferred (small) linux distro for audio?

Hi folks,
   I'm looking for recommendations on a preferred (small) Linux distro for 
surround work. To start with, I'd like to run on a Asus 35 M1-M Pro motherboard 
as I have one handy. Unfortunately, my current Ubuntu distro seems to have 
difficulties picking up its built-in 8 channel audio but my relatively poor 
knowledge/experience of Linux means I can't be sure if I'm doing something 
wrong 
(most probable scenario) or if it's a distro or hardware limitation. As I only 
went for Ubuntu because I had some experience with it already, I thought it 
would make sense, before going further, to seek advice about optimum-for-audio 
distros and concentrate on one of those and preferably one without much bloat.

Dave

--  These are my own views and may or may not be shared by my employer
/*/
/* Dave Malham  http://music.york.ac.uk/staff/research/dave-malham/ */
/* Music Research Centre  */
/* Department of Musichttp://music.york.ac.uk/;*/
/* The University of York  Phone 01904 322448*/
/* Heslington  Fax   01904 322450*/
/* York YO10 5DD */
/* UK   'Ambisonics - Component Imaging for Audio'   */
/*http://www.york.ac.uk/inst/mustech/3d_audio/; */
/*/

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Re: [Sursound] Chasing flies with ambisoinics?

[Eric Benjamin]

 Is there a surround sound method that will 
 reproduce actual depth enough so that you
 could track the movment of a fly in a room? 


A while back I started making a series of simultaneous binaural and 1st-order 
soundfield recordings.  The purpose is to compare them in reproduction, with a 
couple of goals in mind.  I'll leave those goals aside for the moment.  The rig 
is comprised of a dummy head constructed according to ITU-T P.58 and either a 
DPA-4 (for fixed recording situations) tetrahedral microphone or a Tetramic 
(for 
mobile recordings).  A necessary intermediate step has been to perform a 
diffuse-field calibration of both the soundfield microphones and the dummy 
head. 
 That has now been done.  I made need to do it again, but at the moment I have 
something that seems satisfactory.

One of those recordings was of a fly buzzing around the manikin.  As soon as 
Ambisonia is up and running again I'll post some of those recordings.

Eric Benjamin
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Re: [Sursound] Why do you invert the phase of one channel of multi capsule microphones ?

[Eric Benjamin]

 that the axes run along the bisector between 
two of the capsules, or along the trisector between three of the capsules.  
That 
makes a difference in what you call 'front', of course, but also in the 
behavior 
of the array at the highest frequencies.

I hope this helps.

Eric Benjamin



- Original Message 
From: Augustine Leudar augustineleu...@gmail.com
To: sursound@music.vt.edu
Sent: Thu, May 24, 2012 2:57:44 AM
Subject: [Sursound] Why do you invert the phase of one channel of multi capsule 
microphones ?

Hello all,
I am building a six capsule ambisonic microphone. I have been told
that with the opposite capsules (ie up/down, left/right,
forward/backwards) I should invert the phase of one of the channels
and then add them to get the X,Y,Z for the ambisonic b format. I've
been struggling to find a good explanation -  I was wondering if
anyone could explain why this is in detail ?
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Re: [Sursound] Waveplayer - 16 chnl SD-card audio device

[Eric Benjamin]

That is a pretty cool project.

Having said that, I don't know what I would do with one.  As it stands it's a 
2-channel player, but it's possible to get a 2-channel player for less than 
that.  If you expand it to 8 or 16 channels, then it's unique, but I still 
don't 
know what I would do with it.  I suppose that if I were to go to someone's home 
or facility and they  had 8 or 16 speakers but no computer hooked up to them, 
then it might be a good way to interface with that system using a minimum of 
hardware.

Now, if it were able to right to the flash instead of reading from it, that I 
would find interesting.  An 8-channel pocket recorder.  I need one of those.

Eric


- Original Message 
From: Jan Jacob Hofmann j...@sonicarchitecture.de
To: Sursound List sursound@music.vt.edu
Sent: Fri, April 20, 2012 11:54:29 AM
Subject: [Sursound] Waveplayer - 16 chnl SD-card audio device

Dear list,

I was pointed to a device, which is able to play audio-files of up to 16 
channels without the need of a computer as a stand-alone device. It is 
basically 
a SD-card player and I wonder, if this might be interesting for some on this 
list. The price is about  200,- Euro, which makes it really affordable. The 
file 
has to be written onto the SD-card by the use of a computer and a usual 
card-reader/writer, though.
The link is here:

http://www.waveplayer.de/

I even talked to the person developing it and he said, that higher sample rates 
for multichannel might be featured in one of its next software-updates, as this 
would be no technical problem to provide this.

I am quite curious what people on the list might think about this device and if 
there are even experiences or test results about it.

All the best,

Jan Jacob


sound | movement  |  object |  
space
sonic architecture   |site: http://www.sonicarchitecture.de
spatial electronic composition  |  higher order ambisonic music

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Re: [Sursound] Can anyone help with my dissertation please?

[Eric Benjamin]

Robert,

Lots to comment on here.  I seem to be compelled to address your negative or 
not so good observations:

 Not so Good 2) Because one- point miking ignores transient time
 of arrival differences as such , one of the basic cues of sonic perception
 is suppressed explicitly 

That's not really true.  I'm assuming that when you speak of time of arrival 
differences that you are referring to ITDs.  The thing to remember here is that 
ITDs are a function of our presence in the acoustic field, and as such aren't 
present in the recording environment and thus shouldn't be recorded.  In a 
recording and reproduction scenario the ITDs happen in the reproduction of the 
recording, and as it happens ITDs are reproduced very well by Ambisonics, even 
first order Ambisonics.  I showed this quite clearly (I hope) in AES preprint 
8242.  


 3) Impractical number of speakers needed really to work
But one of the really cool things about Ambisonics is that it scales extremely 
well so that it works well with one speaker or two, although not creating 
surround with so few speakers.  And it works quite well with only four 
speakers.  And nowadays there are quite good decoders that work well with ITU 
5-channel arrays.  If higher order sources are available then they can be 
decoded in such a way that the directional resolution is high in the forward 
direction where there are relatively many loudspeakers and not so well to the 
rear where there are relatively few loudspeakers.  


 4) Impractical number of channels needed to really work
Again, that's not really true.  Most common audio carriers have the capability 
to carry many channels, DVD, BluRay.  And many systems are file-based and as 
such aren't really limited at all.  With a system that is inherently 
hierarchical, as Ambisonics is, a broadcast or distrubution system can transmit 
as many or as few channels as is wished.  


 5) In practice, keeping noise low enough is difficult
I'm not entirely sure where this comment comes from.  In terms of natural 
recording, which is what you and I would do but not most of the rest of the 
audio world, the Soundfield microphone as embodied in the Soundfield MkIV and 
MkV microphones, is really quite quiet.  Not as quiet as some modern 
microphones, some of which have self-noise in single digits, but somewhere in 
the mid-teens of dB SPL.  I can't bring to mind any instance when listening to 
the recordings of my colleague Aaron Heller that I was ever aware of the 
presence of noise.  And there's no reason why higher order systems can't be 
made 
very quiet indeed.  Gary Elko mentioned, during the discussion of the MH 
acoustics Eigenmike, that the self noise of the zero order (omni) output is 
about 0 dBA.

So what does my list look like?

Good:
1) Isotropic behavior.  Ambisonics is really good at capturing and reproducing 
ambient sounds.  These are the sounds that inform me that the sound scene had 
some real origin.
2) Reproduction of correct timbre.  While it is relatively easy (but not 
frequently done!) to capture sound with the correct spectrum, 2-channel stereo 
distorts that spectral accuracy in reproduction.  Ambisonics is much better 
although it still suffers from some of the same problems. 

3) Requires lots less speakers than Wave Field Synthesis.

Not so good:
1) I frequently find that I have front/back confusion.

Let the debate continue.


- Original Message 
From: Robert Greene gre...@math.ucla.edu
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Sun, April 1, 2012 8:03:44 PM
Subject: Re: [Sursound] Can anyone help with my dissertation please?


OK I thought that was a good idea, for people to say what they thought
was good and not good about Ambisonics. So here I go(first I guess
but my mother always said Act in haste, repent at leisure. I think
she meant it as cautionary but I have always taken it as advisory!).

Good
1 Elegant as mathematics
2 Forces people to use one point miking which in itself
is already a HUGE thing because it eliminates the absurd
manipulativeness of much of commercial recording practice.
3 In principle, has the capability of reconstructing the complete
soundfield.
4 Puts height in the picture and gets rid of the sound through
a horizontal slit of stereo(which is ironically more like that the better it is 
done!)
5 In practice, more robust than one might have expected
at working over a large listening area (if that matters).
6 In principle, the timbre errors of stereo arising from around the head 
summation are eliminated.

Not so good
1 Emphasis on homogeneity makes it inefficient when not high order.
(Everyone knows that perception to the side of a listener is quite different 
from perception frontally, but this is ignored)
2 (related to 1) Because one- point miking ignores transient time
of arrival differences as such , one of the basic cues of sonic perception
is suppressed explicitly and is only returned to the picture with higher order.
3 

Re: [Sursound] Transient time differences

[Eric Benjamin]

I can't answer the question precisely without either doing an experiment or by 
doing many hours of calculations.  But one thing to consider is that the 
Blumlein recording won't fail to produce correct ITDs at frequencies above 700 
Hz.  At least, not theoretically.  I can demonstrate by calculations using a 
good head model that the ITDs continue to be correct up to some relatively high 
frequency, say up to 6 kHz.  I have the calculations already done to 
demonstrate 
that is works for an Ambisonic system.  


But one of the differences between theory and practice is that the listener 
won't necessarily be exactly at the sweet spot.  That is to say, if the 
listener 
shifts 10 cm to the left then he has undone the 'correct' time differentials 
provided by the ORTF system.  


Don't take this to mean that I don't like ORTF recordings.  I do like them.  
The 
best stereo recording that I have ever made was an ORTF recording. But then, 
I'm 
not a very good recording engineer.  I think that one of the reasons that I 
like 
ORTF is that it introduces an artificial spaciousness which may compensate for 
the spaciousness that is lost in stereo reproduction.

I will do some calculations on ORTF stereo so that I can understand it better.



- Original Message 
From: Robert Greene gre...@math.ucla.edu
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Mon, April 2, 2012 1:44:28 PM
Subject: Re: [Sursound] Transient time differences


Thanks for the information.
But here is my question in more precise form:
Suppose you do a recording with ORTF(which
of course has its own set of problems).
Suppose you record a source that is say 15 degrees
left of center. and that the source is a pistol shot(an impulse).
Now the impulse will arrive at the left mike before
it arrives at the right mike. The time difference
is the same more or less as it would be for
a dummy head recording since the distance between
the mikes is the same (more or less) as the distance
between the listeners ears(or the dummy head ears).
On playback, the impulse will also arrive at the
left ear the same amount of time before the right ear--
as far as the high frequencies are concerned. Namely
they are heavily shadowed  by the head so that the arrival
at the left ear first is blocked from the right ear,
and the right ear hears only the right speaker.
This is in the highs.

Below around 700 Hz, Blumlein would have put the
phase shifts right and that part of time would
be there.

But the head shadowed part , the high frequency
part, is right for ORTF but wrong for Blumlein--
the direct arrival of the high frequency part of
the impulse as recorded in Blumlein is simultaneous
in the two speakers(as is everything) but there
is no reconstruction via head effect because the
head effect is essentially total shadowing.

Of course there is some head shadowing in the real
world, too. But 15 degrees off to one side is not
enough to block the highs so completely as  45 degrees
(or 30 degrees).

So there is some range of angles where the timing
is off because of the greater shadowing(almost complete)
from the wide speaker separation is not representing
the real situation.

Is this wrong? This is not my private theory.
I think Blumlein was aware of this, and I
known other people have mentioned it.

Maybe is does not really matter, but it seems
real enough.
(I believe it is known that time delays in the
high frequencies play a role in location)

Robert

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Re: [Sursound] 32 (ish) channel soundcard for wavedield synthesis ?

[Eric Benjamin]

 currently thinking of using PCI or PCI express cards 
Any time that there are multiple devices the chances of having them be offset 
in 
time increases dramatically.  I've really had a lot of bad experiences, both 
with multiple PCI audio devices and with multiple external ADAT devices.  I'm 
thinking of one system I saw a few years ago that had three ADAT cards in the 
computer to get 24 channels of output.  In that system I occasionally saw 
offsets of greater than a millisecond!  But more recently I've seen a system 
with four MOTU 24I/O devices for 96 channels of output, and that system had 
perfect synchronization between devices.

And of course ASIO drivers are a must.

Eric



- Original Message 
From: Augustine Leudar gustar...@gmail.com
To: sursound@music.vt.edu
Sent: Fri, March 23, 2012 5:11:22 PM
Subject: [Sursound] 32 (ish) channel soundcard for wavedield synthesis ?

Hello all,
I am looking to build a wavefield synthesis setup. I am looking to use
about 32 channels more or less.
I am currently thinking of using PCI or PCI express cards as they a reputed
to be faster than USB/Firewire. The sound cards and drivers need to be
relatively stable as it will be playing 32 channels at the same time.
I am aware of Motus stuff but was wondering if anyone knew of any cheaper
solutions ? I was wondering if there were any ADAT only output sound cards
- I could then fit some Behringer DA converters, Someone also suggested a
M-Audio Lightbridge but its firewire and I dont know how stable the drivers
are = could it handle 32 channels simultaneously  ?
any suggestions appreciated,
cheers,
Augustine
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Re: [Sursound] Decoding coefficients for non symmetrical setups

[Eric Benjamin]

 Bruce Wiggins's (I hope) research was what started this fray out in the first 
place
Yup.  And several others.  But the point is that there is a good deal more to 
be 
done, especially as you point out that:

 this sort of optimization retains the blackbox leanings of machine learning 
 as 
a general discipline
Which would be OK, if the black box could give a meaningful rating of which 
decoders are good and which are bad, or more to the point, which is better than 
another.  But we're not to that point yet.

 how many actually take a look at the early bispectral model of Gerzon?
Took a look at.  But that's not the same thing as implementing it!

On the side of improving the psychoacoustic models I've been working on using 
spherical head models to predict the localization cues achieved and making in 
situ measurements of the ear signals of a real listener when listening to 
Ambisonic reproduction.  Some of this is available in:
Why Ambisonics Does Work , Benjamin, Lee and Heller, AES preprint 8242 (2010)

a paper which was semi-humorous but which also contains some good stuff.  I was 
partly unsuccessful at showing the relationship between Gerzon's Energy vector 
and ILDs and that is something which I will devote some further serious 
attention to soon.

 a more well-thought out optimization criterion, with some intelligent, 
psychoacoustically minded regularization built in snip could still cut the 
mustard
Ah, if only we could find some intelligence to apply to the problem.

Eric


- Original Message 
From: Sampo Syreeni de...@iki.fi
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Wed, February 29, 2012 2:19:46 PM
Subject: Re: [Sursound] Decoding coefficients for non symmetrical setups

On 2012-02-29, Gregory Maxwell wrote:

 Would an automated “blind search algorithm possibly
 
 Speaking of that, you probably want to search the list archives for a thread 
 I 
started in 2009 titled:
 
 A stupid optimizer for irregular ambisonic layouts
 
 In it I provide the source for a simplistic decoder that uses a generic open 
source blackbox non-linear optimizer library with a simple objective to make 
matrixes.

Before They point it out themselves, I think the fourth installment of Blah 
does 
very much the same. And of course Bruce Wiggins's (I hope) research was what 
started this fray out in the first place. So, yes, this is something that seems 
to be recommended from more than one corner, with regard to irregular layouts. 
But still...

Personally what I find a bit worrisome is that this sort of optimization 
retains 
the blackbox leanings of machine learning as a general discipline. None of the 
ambisonic specific, closed form optimization literature, or the derived 
specifics of the base optimization problem, are being utilized. Instead the two 
(sometimes simultaneous, sometimes even not that) Gerzonian equations are being 
fed into one or another optimization framework, with no regard to what happens 
then, and without feeding in all of the age-old mathematical-physical knowhow 
of 
how those systems of equations behave. Like for instance psychoacoustical 
sensitivity estimates from the BBC era.

In addition to being a fan of black box algorithms, including all of the stuff 
that goes under the rubric of data mining (professionally I make my living as 
a database guy), I'm also a little bit of a skeptic towards the stuff. At least 
as far as the math I know and love suggests I should be.

For example, when using support vector machines to fit polynomial bases, how 
many people actually care to evaluate the Vapnik-Cervonekis bound intrinsic to 
the problem, and then bound it in a principled fashion before commencing to 
optimize numerically? That after all is the most principled framework in which 
to bound overfitting by the machine -- i.e. the very same thing which leads to 
speaker detent within the ambisonic framework, even after simple dimensional 
constraints have already been dealt with.

And how many actually take a look at the early bispectral model of Gerzon? Or 
the third one which name I don't remember right now? Even if those aren't 
backed 
up by psychoacoustics, they are still very, *very* relevant as (easily, 
formally, in-principled-fashion) saturable optimization criteria (in the usual 
ambisonic L^2 sense no less).

I don't think going with the easy route and just using blackbox optimizers does 
the job best, here. Instead, I would think we have to find a way to inject more 
and more current, analytically purified, psychoacoustic knowledge into the 
system, before we even start to optimize. Even if numerical optimization still 
remains the key in reaching a local optimum in this kind of a very difficult 
nonlinear optimization problem.

Once again, Robert Greene, please help me if I'm falling short on the hard 
math, 
somehow.

 I like the generic optimization approaches _more_ than more mathematically 
elegant closed form solutions because it's easy to play 

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

[Eric Benjamin]

Robert Greene wrote:
 there are VERY serious problems of other kinds with using it at the kinds of 
distances 

 (fractions of a meter less than 1/2 , much less often enough) where proximity
 effect becomes really major.

That is indeed true, except perhaps for the label of Very.

I first noticed this in making measurements of soundfield microphones, not in 
analysis.  My measurement of the 'W' response showed proximity effect.  I 
believe that there are two reasons for this, both having to do with 
non-coincidence.  We model the outputs of the soundfield microphone array as 
'W', the sum of all the capsules (in the typical case) and X, Y, and Z, the 
differences between pairs of capsules.  These correspond to a coincident set of 
a monopole and three orthogonal dipoles at the center of the array.  We know 
that that model is not exactly accurate, especially in the high-frequency 
case.  
At frequencies where the distance across the array is a significant fraction of 
a wavelength there is a significant phase difference between the various 
capsule 
signals with the result that the frequency responses of the array change, and 
that they also are direction dependent.  One way of looking at that result is 
that the free-field and diffuse-field frequency responses begin to differ from 
each other at high frequencies.  This really happens fairly abruptly, with the 
diffuse-field omni response rolling off quite rapidly above 10 kHz for 
soundfield arrays with the typical 1.47 cm radius.

This is not a good thing.  What makes it tolerable is that the diffuse-field 
response of even a small (1/2) omni starts to roll off at just a few kHz.  By 
that measure a soundfield microphone is quite a good !

Likewise, we wish that the behavior of a soundfield microphone array were ideal 
at low frequencies, but it's not.  depending on the direction of arrival some 
of 
the capsules will be nearer to the source than others, so the level and the 
boost due to proximity effect is greater for the near capsules than for the far 
capsules.  I believe that this effect is more significant for 'W' than for the 
dipole outputs, basically because we expect the dipole outputs to have 
proximity 
effect but not the monopole output.

We should keep in mind that the near-field behavior of even conventional 
monopole (pressure) and dipole (figure eight) microphones is not ideal.  As I 
mentioned above, the DF response of omni microphones rolls off quite early.  
The 
DF response of figure-eight microphones tends to be a bit better, although 
still 
not ideal.  Also, the nearfield properties of conventional figure eight 
microphones isn't ideal.

If one were to need a soundfield microphone with ideal directional properties, 
it seems as though the only option is something like the Microflown.  But that 
has it's own set of problems.


- Original Message 
From: Robert Greene gre...@math.ucla.edu
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Sat, July 23, 2011 5:45:48 PM
Subject: Re: [Sursound] the recent 2-channel 3D sound formats and their 
viability for actual 360 degree sound


I feel a little diffident in commenting on this in the presence of so many 
experts on the Soundfield mike in theory as well as in practice,
but unless I am misunderstanding how it works, there are VERY serious problems 
of other kinds with using it at the kinds of distances (fractions of a meter 
less than 1/2 , much less often enough) where proximity
effect becomes really major.

Namely, as I understand it, the way the B format signals are built is 
predicated 
upon the distances among the four capsules being quite small
compared to the distance of the source, for the following reason:
Compensation is needed for the fact that the capsules are on the faces of a 
tetrahedron, not coincident and all at the center. This compensation
is based on the fact that at reasonable distances to the source, the 
differences 
of the distances to the mikes is obtained by orthogonal projection on the axis 
of arrival of the sound(to a very good apporximation).

To make sense of this jargon, suppose a source is on the line that is equistant 
from three of the capsules.  Then its distance to those three
will always be the same, and if the source is reasonably far away the distance 
to the fourth capsule will be a constnat. This comes from the Pythagorean 
theorem limit case in effect: at large distances , the
difference between A to S and B to S is equal to the length of the projection 
of 
the line from A to B onto the line from A to S (or B to S these being parallel 
in the limit case).

If one does NOT have such large distance to the source, the variation of 
distances to the capsules will be extreme and also complicated.
Just think of how the distances to the four face centers of the tetrahedron 
will 
vary in odd ways when the source is close by!

So it seems to me(and I am prepared to be all wrong!) that
the Soundfield mike could not be expected to 

Re: [Sursound] Opinions on the Brahma soundfield mic kit

[Eric Benjamin]

Dan Andrew
 whats the catch?

I'm not sure that there is a catch, as such.  

It's apparent that the Brahma is an 'A-format' microphone, in which it is the 
capsule signals that are recorded and not the Ambisonic B-format.  To get 
B-format will require some matrix processing (sum and difference of capsule 
signals), and some equalization to restore flat frequency response.  Not only 
that, but the microphone array appears to be made up of omnidirectional signals 
which means that the difference signals (B-format X, Y, and Z) will need to 
have 
the low frequencies boosted substantially to give any semblance of flat 
frequency response.

Perhaps the Oomagamma folks can supply some commentary, and much better than 
that, perhaps they can supply some A-format demonstration files.




- Original Message 
From: Dan Andrews d...@db-av.co.uk
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Mon, July 11, 2011 4:36:34 AM
Subject: [Sursound] Opinions on the Brahma soundfield mic kit

Im looking to buy a mic for b-format recording and wondered if anyone on the
mailing list has had any experience with Brahma soundfield mic kit?

http://www.oomagamma.com/brahma_kit/brahma_kit.html

This kit includes the mic, a modified 4ch Zoom 24/96 recorder, cables, a
shockmount, 2 wind shields and a wooden case to put it all in, all for 729
euro

This all seems far to good to be true, whats the catch?

All the best

Dan


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

[Eric Benjamin]

 all of my material that was on there (plus a bit more), and all of John 
Leonard's,
I hadn't visited your pages in a while.  I particular like the roll-over 
informational photos.

Between your material and John Leonard's material you have Early Music and 
environmental sounds fairly well covered.  Now we need some more variety!

I'll contact you off-list.

Eric



- Original Message 
From: Paul Hodges pwh-surro...@cassland.org
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Mon, July 11, 2011 12:45:07 PM
Subject: Re: [Sursound] the recent 2-channel 3D sound formats and their 
viability for actual 360 degree sound

--On 10 July 2011 22:47 +0200 Jörn Nettingsmeier netti...@stackingdwarves.net 
wrote:

 the demise of ambisonia.com is lamentable,

Indeed.  But I'd like just to remind people that all of my material that was on 
there (plus a bit more), and all of John Leonard's, and Richard Lee's articles, 
are now available from my site here: http://ambisonic.info/audio.html and 
here: http://ambisonic.info/info.html.

I will make similar pages for anyone else's stuff if they ask me to.

Paul

-- Paul Hodges


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

[Eric Benjamin]

Rober Greene wrote:
 There was a method developed by Finsterle 

Tell us more about it.  Is the method described elsewhere?  Is it embodied in a 
device, or software?  Who is Finsterle?

Eric



- Original Message 
From: Robert Greene gre...@math.ucla.edu
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Sat, July 9, 2011 8:22:04 PM
Subject: Re: [Sursound] the recent 2-channel 3D sound formats and their 
viability for actual 360 degree sound


There was a method developed by Finsterle that worked very well
indeed, much better than Trifield(which has always seemed to me
to have a serious center detent.
Finsterle's method  had sound in the rear psychoacoustically
encoded not to sound in the rear but to solidify the front
images.
This worked very well in my experience
Robert

On Sat, 9 Jul 2011, Paul Hodges wrote:

 --On 09 July 2011 14:04 -0400 Marc Lavall?e m...@hacklava.net wrote:
 
 So, is it possible to adapt a stereo recording to play on a horizontal
 ambisonics system, in order to get a better stereo image than with
 conventional stereo? A kind of restored stereo experience that
 ambisonics can provide because of its directional capabilities?
 
 Two approaches that Michael Gerzon took are exemplified by the Super Stereo 
mode of the early ambisonic decoders, and the later Trifield system using 
three speakers; but neither of these is about attempting to generate a full 
circle from the stereo signal.  A problem that arises, in any case, is that 
the 
result does depend strongly on the way the stereo recording was made - 
coincident mics (e.g. Blumlein), spaced mics (e.g. Decca Tree), or a reliance 
on 
mixing from spot-mics.  As these record very different directional cues, a 
single process can't be expected to handle them all equally effectively.
 
 As for 5.1 - there are a number of useful decoders available which can be 
 used 
to reproduce ambisonic signals using speakers set up for 5.1; but the 
irregular 
spacing means inevitably that the results are not as good in some directions 
as 
they could be with the same speakers more uniformly spaced. Playing 5.1 
signals 
through an ambisonic system is a matter of steering those signals as virtual 
sources at the required angles in a B-format signal; as with stereo, nothing 
is 
added to the experience because there is nothing extra to be found - but the 
reproduction will be less good to the extent that the sources expected when 
the 
5.1 mix was done are being less precisely reproduced.
 
 Paul
 
 -- Paul Hodges
 
 
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Re: [Sursound] B format mic using omnis?

[Eric Benjamin]

 building/using a soundfield type mic using omni's?

If life gives you 4060s, then make lemonade.  I mean, a B-format microphone!

The problem gets a lot easier if you resign yourself to making something that 
will have good utility as opposed to making something optimal.

I would make a microphone array using a spherical baffle.  One can find quite a 
variety of wood spheres.  Here in the US there are spheres made of Birch with 
diameters of 1-1/2, 2,   Wood is great because it's cheap, easy to drill, 
and if you make a mistake you just grab another one.  


If the user doesn't intend to make use of height, I'd want to make the array a 
horizontal-only one, primarily because the drilling is a lot easier!  It's 
difficult enough to find the equator of a sphere without having to find the 
vertices of a tetrahedron inscribed in the sphere!

The choice of diameter is tough, because as previous respondents pointed out 
there is a tradeoff between SNR and bandwidth.  As you know, the 1st order 
patterns will be derived by subtracting the outputs of 2 or more of the 
capsules, and that means that the response will have to be equalized by apply 
an 
LF boost below a critical frequency determined by the diameter of the sphere.  
For an open array this is straightforward but for a spherical baffle you need 
to 
include the diffraction of the sphere.  I can calculate this, but not on the 
back of an envelope.  The spherical diffraction gives an effective gain of 6 dB 
and this is worth going before because the self noise of the 4060s is about 23 
dBA as I recall, which is good enough to be useful but not so generous as to 
allow one to easily throw it away.  So what we would like to do is to have that 
critical frequency be somewhere near the frequency at which the ear is most 
sensitive to mic hiss - about 2 to 7 kHz.  And typical usable sphere sizes just 
happen to do that.  This means that the array will only work well up to about 
10 
kHz, but then that is true of a traditional SF microphone too!

The construction may be just a little bit difficult.  It turns out to be 
difficult to find the center of a sphere once you have it in hand.  You will 
really need to use a drill press to drill the holes.  Routing the microphones 
into the sphere will also be difficult, depending on how the end of the 
microphone cables are connectorized.  It may turn out that you will want to 
drill a large hole in the sphere at a direction not populated by microphone 
capsules, and use that hole for entry of the microphones and to route them each 
into their respective holes.

Having done this before, I can give you a bit more specific info if you contact 
me off-list.

Eric Benjamin



- Original Message 
From: Dave Malham dave.mal...@york.ac.uk
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Mon, June 20, 2011 3:54:38 AM
Subject: [Sursound] B format mic using omnis?


May seem a strange question, but anyone ever had any experience of 
building/using a soundfield type mic using omni's? I have been asked by one of 
the artists featured on The Morning Line if there's anything he could do with 
his collection of 4 DPA's (4060-bm's). Not something I'd ever really though 
about before, but as Angelo's B format hydrophone uses omni's ... 
(http://www.angelofarina.it/Public/UAM-2011/)

    Dave

--  These are my own views and may or may not be shared by my employer
/*/
/* Dave Malham  http://music.york.ac.uk/staff/research/dave-malham/ */
/* Music Research Centre                         */
/* Department of Music    http://music.york.ac.uk/              */
/* The University of York  Phone 01904 432448                        */
/* Heslington              Fax  01904 432450                        */
/* York YO10 5DD                                                    */
/* UK                  'Ambisonics - Component Imaging for Audio'  */
/*                    http://www.york.ac.uk/inst/mustech/3d_audio/; */
/*/

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Re: [Sursound] Jacktrip (bandwidth)

[Eric Benjamin]

As a (mostly) non-user of Linux, I'm uncertain as to what Jacktrip is used 
for.  
Is it to stream data over a network?  Or is it to distribute computing between 
multiple machines?  If it is the former, then I'm puzzled.  I can easily play a 
16-channel file hosted on one machine on my 100 baseT network from another 
machine on the network.  That only takes about 13% of the network bandwidth.  I 
can stream the data for hours without dropping a single sample.  I can't report 
on the ability to stream larger numbers of channels because I don't have the 
hardware to do it.  Or the need, for that matter.

Eric



- Original Message 
From: Michael Chapman s...@mchapman.com
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Wed, June 8, 2011 3:02:03 AM
Subject: [Sursound] Jacktrip (bandwidth)


I can't find any indicative performance (bandwidth)
figures for Jacktrip ... so ask for the experience of
others.

On a standard CAT-5 cable between two adjacent
machines I can get four (mono) channels at 48 KHz,
but trying to set channels 4 just results in a (very
silent) failure to connect.

Back of an envelope calculations of audio flux
against 100 Mb/s (say 10 MB/s) suggest more
should be possible.
That said secure copy (scp) of files seems to
run at 100Mb/s.

Anyone done better ?

Michael
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Re: [Sursound] Minim AD7 for sale - Speaker configs.

[Eric Benjamin]

 from psychoacoustics we cannot really hear directions of sound below 80 Hz
I know that it is frequently written, but it's not true. Of the two 
localization 
mechanisms active at low frequencies, Interaural Time Differences (ITDs) are 
the 
ones that give useable localization cues in free space.  The time difference 
depends only on the direction of the source and not on the frequency.  
Unofrtunately there is very little information in the psychoacoustical 
literature about low-frequency localization.  They consider 250 Hz to be 'low' 
and 100 Hz to be very low.  Maybe some day I'll do some research on that...

It is true that the threshold of hearing rises substantially at low 
frequencies, 
and for that reason localization acuity decreases.

Here's what I think really happens.  For low frequency sounds reproduced in 
ordinary rooms, the first arrival at the listener's two ears naturally has ITDs 
that correspond to the direction of the source.  After a short period of time, 
reinforcement of the sound by reflections from the room boundaries changes the 
phase of the sounds at the ears.  This can be more easily seen by considering 
the modal structure of the room at low frequencies.  The room has relatively 
few 
modes and the sound wave quickly becomes constrained to travel in the modes.  
Because of the relative energy of the transverse, oblique, and tangential 
modes, 
the sound effectively comes from the direction of the mode, not of the 
source.  In practice, large ITDs AND ILDs are seen at the listener's ears.  As 
a 
result, the percept will probably be that the sound is coming from a direction 
other than its actual source.  This is what I actually hear when using low 
frequency test signals in real rooms.

But there's more going on that that.  Almost always, the low-frequency sound 
has 
actually a fairly broadband spectrum.  With that sort of signal the auditory 
system clearly evaluates several cues as to the source direction and gives a 
best estimate of the actual direction of the source.

There are good reasons to use several subwoofers in a multichannel reproduction 
system.  At least the following two papers support that idea.

[1] Subkey, A., Cabrera, D., Ferguson, S.; Localization and Image Size Effects 
for Low Frequency Sound, AES preprint 6325 (2005 May)
[2] Martens, W., The impact of decorrelated low-frequency reproduction on 
auditory spatial imagery: are two subwoofers better than one? presented at the 
AES16th International Conference, Rovaniemi, Finland, (1999 April)
 
Having said that, I only have one (large!) subwoofer in my multichannel 
listening room.  But the reason for it has more to do with $$$ than my 
believing 
that one is enough.
 
Eric


- Original Message 
From: Bo-Erik Sandholm bo-erik.sandh...@ericsson.com
To: Surround Sound discussion group sursound@music.vt.edu
Sent: Wed, May 4, 2011 4:41:02 AM
Subject: Re: [Sursound] Minim AD7 for sale - Speaker configs.


Yes, I have a few woofers available BUT according to what I understand
from psychoacoustics we cannot really hear directions of sound below 80 Hz,
as the ear/brain is changing method of decoding soundwaves between 80 to 100 Hz?
So do I really need more than a pair driven in mono (or 4) to even out the 
excitation of the room modes? 


If I where to add a low frequency decoder how should I do that?

Is it not so that the speaker feed to all of the 10 speakers are in phase for 
frequencies lower than 

Some undefined frequency? 
That is using Ambdec?

Should I have a highpass filter before or after the decoder for the 10 small 
speakers,
if I add a low frequency feed, either mono or decoded?

Bo-Erik

-Original Message-
From: sursound-boun...@music.vt.edu [mailto:sursound-boun...@music.vt.edu] On 
Behalf Of Peter Lennox
Sent: den 4 maj 2011 11:32
To: Surround Sound discussion group
Subject: Re: [Sursound] Minim AD7 for sale - Speaker configs.

Quick suggestion: - as you're having to use more than 8 channels anyway, you're 
likely to be using a 16 channel card; thus, you would have some channels left 
to 
decode (horizontal only) to 3 or 4 subs

Dr Peter Lennox
School of Technology
University of Derby, UK
tel: 01332 593155
e: p.len...@derby.ac.uk  


-Original Message-
From: sursound-boun...@music.vt.edu [mailto:sursound-boun...@music.vt.edu] On 
Behalf Of Bo-Erik Sandholm
Sent: 04 May 2011 08:09
To: Surround Sound discussion group
Subject: Re: [Sursound] Minim AD7 for sale - Speaker configs.



I still want to suggest a setup that I will soon have in operation, I have 
written about it before.
It uses 10 channels, it is a hexagon in the horizontal plane with a speakers at 
front back.
The Z is handled by for speakers, placed where the 4 hexagon side speakers will 
end up if the 

Hexagon is rotated 90 degrees around a axis through the front and back hexagon 
speakers.

Fons A has created a ambdec for me of this setup.

I have the possibility to have all speakers except the 2 floor 

[Sursound] Ambisonics symposium 2011: was help with links ...

[Eric Benjamin]

Peter Lennox p.len...@derby.ac.uk wrote:
 anyone taking something interesting to the Ambisonics symposium 2011 in 
Kentucky?


I'm planning on bring a prototype of a practical, affordable second-order 
soundfield microphone.  Of course it's not done yet, and perhaps I'm talking 
through my hat.  Why would I claim something that I haven't finished?  To put 
the pressure on so that I do finish!

Eric
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Re: [Sursound] umashankar mantravadi has shared documents with you

[Eric Benjamin]

Hmm.  The link is not working for me.  Can you double check it?

Thanks.

Eric



- Original Message 
From: umashankar mantravadi umasha...@hotmail.com
To: sursound@music.vt.edu
Sent: Mon, November 8, 2010 9:19:32 PM
Subject: [Sursound]  umashankar mantravadi has shared documents with you

umashankar mantravadi shared the folder ambisonics with you on Windows Live.
http://cid-3aeeea022e2ad294.profile.live.com/?Bsrc=EMSHOOBpub=SN.Notifications

a few files done with a 6 mm tetrahedron. diwali crackers, but not a very 
impressive show.
View folder
http://cid-3aeeea022e2ad294.skydrive.live.com/redir.aspx?page=browseresid=3AEEEA022E2AD294!204type=6authkey=9imRr2LuD0M%24Bsrc=EMSHOOBpub=SN.Notifications

Notifications preferences
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Microsoft privacy statement
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