I think I may be having a microphone epiphany or at least what smug boffins
might call "an intuitive understanding" but please feel free to correct any
errors in my thinking. In using a six capsule (or 3 capsule) microphone
setup with 3 figure of eight pairs this gives you localisation information
based on the pressure gradient between each of these pairs (given by the
antiphase ?) . This pressure gradient gives you the localisation of the
sound on any of the given axis (X,Y,Z) and with this information you can
get the localisation of the sound in a sphere ? Is this correct , or even
vaguely correct ?

On 25 May 2012 18:05, Augustine Leudar <[email protected]> wrote:

> Thankyou all for your responses - they've been a great help - especially
> Eric Benjamin - that makes perfect sense thanks !
>
>
>> There are two ways to approach the construction of such a microphone.
>>  One of
>> them is to use cardioid microphone capsules facing outwards at the six
>> sides of
>> your cube.  The other is to use omnidirectional capsules and place them
>> on the
>> surface of a rigid object such as a sphere.  I don't know which approach
>> you're
>> taking so I'll discuss both.
>>
>> Using a cubic array is intuitively simple for a microphone that is going
>> to have
>> outputs which are directed along the Cartesian axes, such as is the case
>> for
>> Ambisonic B-format.  As you already know, in B-format the W signal is
>> equivalent
>> to an omnidirectional microphone located at the center of the array.  X
>> is a
>> dipole or figure-of-eight microphone pointed along the X-axis.  Since a
>> dipole
>> has two lobes, one positive and one negative, it must have a precise
>> orientation
>> to conform with X.  X is a front-back signal, Y is a left-right signal,
>> and Z is
>> an up-down signal.
>>
>> Cardioid microphone capsules have a directivity that is heart-shaped,
>> hence the
>> name.  They theoretically have maximum sensitivity in the front direction
>> and
>> zero sensitivity in the rear direction.  They can be modeled as being the
>> sum of
>> a monopole (omni) and a dipole (figure-of-eight), and that is why it is
>> easy to
>> recover those components by adding or subtracting capsule signals.  If
>> you have
>> two cardioid capsules, on a line and facing opposite directions, when you
>> add
>> the two microphone signals the two dipole components, being of opposite
>> polarity
>> to each other, cancel out.  That is assuming that the microphones are
>> identical,
>> of course!  If you subtract one of the microphone signals from the other,
>> the
>> two monopole components, being identical, cancel out, but the dipole
>> signals now
>> add to each other.  So the sum gives a monopole and the difference gives a
>> dipole.
>>
>> Note that it only takes two of the capsules to get a monopole.  But you
>> will
>> have 6 capsule signals and you can sum all of them to get a monopole
>> signal that
>> is, at least in some ways a better omni than you get with just two.
>>
>> The process is the same, of course, with the other two axes and deriving
>> Y and
>> Z.
>>
>> Now, if your capsules are omnidirectional microphones the analysis is
>> somewhat
>> different.  There really should be some sort of a baffle on which to
>> mount the
>> microphone capsules although it's certainly possible to have them just
>> sitting
>> in free space.  But let's assume that they are mounted on (actually, in)
>> the
>> surface of a sphere.  If we add all six omni capsules together we get
>> another
>> omni.  Simple.  But if two of the opposite-facing omni capsules are
>> subtracted,
>> we get a dipole.  That may not be immediately apparent.  It is the
>> separation in
>> space that makes it so.  The simplest analysis is that, if the two omni
>> capsules
>> are identical, when you subtract the two signals you will get nothing.
>>  And
>> that's partly true.  But since the capsules are separated by some small
>> distance
>> there will be a small difference between the two signals.  If a sound
>> wave is
>> approaching along the line separating the two capsules then it will reach
>> one
>> capsule first and the other one second.  Assuming that the sound source is
>> distant from the microphone array there will be no attenuation in level
>> between
>> the two microphone positions, although there may be some difference in
>> level due
>> to the spherical baffle.  Mostly there is a difference in the phase of
>> the two
>> signals.  When the two are subtracted, one from the other, a difference
>> signal
>> appears that is proportional to the phase difference.  But that phase
>> difference
>> is proportional to the frequency of the sound!  That is, at low
>> frequencies, say
>> 100 Hz, the wavelength of sound is about 3.4 meters.  And high
>> frequencies, say
>> 10 kHz, the wavelength is 3.4 cm.  If our capsules are separated by 1 cm
>> (just
>> to make the arithmetic easy), then that is 1/340 of a wavelength at 100
>> Hz and
>> 1/3.4 of a wavelength at 10 kHz.  So there will be a minute difference in
>> phase
>> at 100 Hz and a large difference in phase at 10 kHz.  The result is that
>> the
>> frequency response of such an array is differentiated; it rises at 6
>> dB/octave
>> from the lowest frequencies to the highest frequencies.  That slope will
>> need to
>> be corrected by equalization.  Furthermore, if the spacing were
>> increased, say
>> from 1 cm to 10 cm, the amount of signal will be increased by a factor of
>> 10.
>>  But a sound with a wavelength of 3.4 cm, like our 10 kHz sound, will be
>> ambiguous.
>>
>> It gets quite complicated.  I have simplified things to make the
>> explanation
>> tractable.  Typical spherical microphone arrays have diameters of about 5
>> to 10
>> cm, depending on what bandwidth is needed in the final design.
>>
>> One other comment; You could use a different orientation of the array
>> than what
>> I have assumed above where the capsules lie on the Cartesian axes.  You
>> could,
>> for instance, rotate the array so 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 <[email protected]>
>> To: [email protected]
>> 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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>> [email protected]
>> https://mail.music.vt.edu/mailman/listinfo/sursound
>>
>>
>>
>> ------------------------------
>>
>> Message: 7
>> Date: Fri, 25 May 2012 08:58:07 +0200
>> From: Bo-Erik Sandholm <[email protected]>
>> Subject: [Sursound] Suitable students project - Creation of ambisonic
>>        player  ?
>> To: Surround Sound discussion group <[email protected]>
>> Message-ID:
>>        <
>> e023323b1ad21d44af70273b35e750151660a05...@esesscms0356.eemea.ericsson.se
>> >
>>
>> Content-Type: text/plain; charset="us-ascii"
>>
>>
>> I got the silly idea that if we have few a students with a lot of time at
>> their hands...
>>
>> It could be possible to create a low cost ambisonic player using the
>> following resources
>>
>> http://flac.sourceforge.net/format.html
>>
>> To encode .amb into, i guess att the moment there is yet no support for
>> http://mchapman.com/amb/wavex in flac.
>> Even if flack encoding is not neccessary but can save storage space.
>>
>> http://dream.cs.bath.ac.uk/researchdev/wave-ex/bformat.html
>>
>> Player
>> http://www.alsaplayer.org/features.php supports FLAC ambisonic player
>> http://offog.org/code/potamus.html or this simple one that i know works
>> well via jack.
>>
>> Ambisonic decoder
>> http://kokkinizita.linuxaudio.org/linuxaudio/index.html
>>
>>
>> Operation system
>> http://bkhome.org/blog/?viewCat=General
>> - Barry Kauler has a A-10 Puppy Linux distribution and development
>> environment, that seems nice - Puppy linux version
>> Not shure jack is included, but jack is available for puppylinux
>>
>> Ubuntu is also available for A-10
>>
>> Below 100 USD computers
>>
>> Computer with 4 Core processor and 8 channel sound card:
>>
>> http://www.aliexpress.com/product-fm/563764893-New-released-Android4-0-IPTV-google-tv-smart-android-box-allwinner-A10-Model-MK802-Notice-just-wholesalers.html
>>
>> Or
>> http://rhombus-tech.net/allwinner_a10/hacking_the_mele_a1000/
>>
>> Mele 2000 from dealextreme cost 6 usd more but still costs below 100USD
>> and has 4 GB flash memory instead.
>>
>> With this platform it is is possible to have up to 8 speaker channels
>> from the computer via hdmi,
>> I have not at the moment a link to a 8 channel power amplifier to use for
>> 2 rings of 4 speakers,
>> But a horisontal 6 speaker array or other commeon speaker configurations
>> can easily be powered via hometheater surround amplifiers.
>>
>> OR
>> Maybe to offer a "novelity" integrate the ambdec decoder and a
>> multichannel wave player and port it to the google TV android environmnet
>> with a number of "suitable" decoding matrixes and publish it together with
>> a pointer to a bit of ambisonic material.
>>
>> Have a nice summer
>> Bo-Erik
>>
>> ------------------------------
>>
>> _______________________________________________
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>>
>> End of Sursound Digest, Vol 46, Issue 7
>> ***************************************
>>
>
>
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