from:"Richard"

Re: [Sursound] New Spatial Music: Ambisonic Music Library

2022-10-07 Thread richard ford

Many thanks for clearing that up, UHJ all the way (can't wear headphones after
years of ear abuse playing in a band)
Will add a link 
Richard

Hi Richard,
Thanks for your email. The options are the following:
£5.00 - Stereo (binaural) files only.£10.00 - Spatial bundle (digital),
including master files in stereo (binaural), first (4ch) and third (16ch) order
Ambisonics (AmbiX ordering). These are delivered via email after purchasing.
You receive all spatial versions of these files within the download options
(including UHJ and Quark decodes, too).
Hope that clears things up!
And regarding a link from your website, no problem at all, thank you for your
support! Feel free to add a link.
Best regardsJames

On 7 Oct 2022, at 12:50, richard ford wrote:
Hi
A quick question, if i may. There appears to be no mention of which version of
the album (ambisonic or binaural) your paying for when goung for the £5+
option.Could you please advise.
Also, would you be adverse to me providing a link to your pahe from my website?
Richard

Dear Sursounders,

I’m pleased to announce the second spatial music record by the Ambisonic Music
Library label, Image Shift’s ‘Anchored Transients’, out today 7th October. The
label has been established to promote and publish music specifically created in
ambisonics.

It’s presented in binaural stereo, first and third order ambisonics (AmbiX
ordering), and available to listen to in full at:

https://imageshift.bandcamp.com/album/anchored-transients
<https://imageshift.bandcamp.com/album/anchored-transients>

If anyone is interested in proposing a release for the label, or has any
questions, please don’t hesitate to get in touch at
i...@ambisonicmusiclibrary.com <mailto:i...@ambisonicmusiclibrary.com>

About:
The debut ambisonic record from Image Shift delivers a focused series of sonic
sculptures, featuring clean electronic lines and spatial rhythms. Deliberate
dynamic shapes shift and evolve within an ever expanding 3D panorama, exploring
space in higher-order ambisonics from the ground up.

'Anchored Transients' draws us in... and hits us right in the chest, expanding
and collapsing the sound field... 'Error' disrupts and misshapes sound
materials, fighting and regaining control with every bar... 'Symbiosis' is
ambisonics for the club, a native expression of electronic house in full 3D
form... finishing with 'The Synth Room', a palette cleanser of pop with tight
transitions and melodies. A great introduction to the world of Image Shift.

Explore in binaural (stereo) via headphones, or via first or third-order
Ambisonics through your own chosen decoding method.

We’d be incredibly grateful for any feedback, and we look forward to sharing
more exciting spatial music with you very soon.

Have a great weekend,
Best regards
James Bagshaw
Head of Things
Ambisonic Music Library
https://ambisonicmusiclibrary.com <https://ambisonicmusiclibrary.com/>
https://ambisonicmusiclibrary.bandcamp.com
<https://ambisonicmusiclibrary.bandcamp.com/>
T: +44 788 291 5002
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Re: [Sursound] New Spatial Music: Ambisonic Music Library

2022-10-07 Thread richard ford

Hi
A quick question, if i may. There appears to be no mention of which version of
the album (ambisonic or binaural) your paying for when goung for the £5+
option.Could you please advise.
Also, would you be adverse to me providing a link to your pahe from my website?
Richard

Dear Sursounders,

It’s presented in binaural stereo, first and third order ambisonics (AmbiX
ordering), and available to listen to in full at:

https://imageshift.bandcamp.com/album/anchored-transients
<https://imageshift.bandcamp.com/album/anchored-transients>

If anyone is interested in proposing a release for the label, or has any
questions, please don’t hesitate to get in touch at
i...@ambisonicmusiclibrary.com <mailto:i...@ambisonicmusiclibrary.com>

Explore in binaural (stereo) via headphones, or via first or third-order
Ambisonics through your own chosen decoding method.

We’d be incredibly grateful for any feedback, and we look forward to sharing
more exciting spatial music with you very soon.

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Re: [Sursound] format convention of Ambisonic Sound Library Files

2022-05-25 Thread Richard Dobson

Welcome to my world - as it used to be! The ring-pass-not is that word 
'standardised". My AMB Wavex-based file format is now 22 years old 
(FuMa, up to third-order). People have long moved on - basically as soon 
as that was published, everyone here started arguing about all the other 
information that should be included, alternative standards, channel 
order, higher orders,  filters, irregular layouts, the works. I am still 
not too sure if even now things are standardised enough for anyone to 
write a fresh file format for it that pleases everyone. Suffice it to 
say, such things can't just be written,they have to be implemented and 
tested, e.g. on vast tiered speaker layouts which very few people have 
access to. And of course it will need to use a 64bit-friendly file 
format too...


A reminder, for anyone feeling nostalgic:

http://www.rwdobson.com/bformat.html

Richard Dobson



On 24/05/2022 23:13, Sampo Syreeni wrote:

On 2022-05-24, Alan Kan wrote:

Ah… it seems one has to actually click on the file to see that detail. 
I was downloading from the list page.


Why not embed the a standardised format descriptor into the file itself? 
Most formats permit that. In fact that's what we've been doing all along 
e.g.in RIFF WAVE.

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Re: [Sursound] Ambisonic UHJ Stereo decoder to speaker feeds

2022-03-05 Thread richard ford

Bruce
Firstly, thank you for the new plug-in, looking forward to trying it out, once 
i het it working.
And a special thanks for the information about the level of 'W' being a little 
'hot'
I did some quick tests of a couple of albums and after reducing 'W' by 3db 
before feeding into your earlier B-Format to Square Wigware decoder and the 
results were certainly sounding more like what was originally promised of the 
system back in the day.
Could this be oe of the reasons why the system neber impressed people if this 
error was never caught when the harware was developed? 
Cheers

  I've released a classic UHJ to square/rectangle decoder which can be found
at: https://www.brucewiggins.co.uk/?p=1836

It can also pass out B-format with or without shelf filtering applied.

It's a JS effect, so Reaper is the ideal platform (there is a JSFX VST that
should work on other DAWs, too).  It implements both the shelf filters and
forward preference controls as suggested in:
Gerzon, M.A., 1985. Ambisonics in multichannel broadcasting and video. *Journal
of the Audio Engineering Society*, *33*(11), pp.859-871.

As an aside, looking through the old papers, it looks to me that the UHJ
equations actually result in a W signal that's 3dB hotter than the SF mic
standard (i.e. AmbiX gains!).  Looking at the wireless world articles also
suggests this (B format input to a unit that implements UHJ applied 3dB
boost to W before decoding).

cheers

Bruce

On Thu, 24 Feb 2022, 23:16 surround surround,  wrote:

> I have retired my Onkyo SV-909 receiver with a UHJ decoder after many
> years of service. Over the years I have transferred my LP and CD surround
> recordings decoded into multichannel flac files for playback on a computer
> based system. I did manage to do a couple UHJ recordings using the Onkyo a
> while back but wondered if their might be a computer based solution.
>
>
> I am currently using Reaper as my DAW. I did a search for Ambsonic VST
> plug-ins and found The Ambisonic Toolkit for Reaper. I played with that but
> found that that The UHJ decoding it could do was from UHJ W,X,Y,Z to stereo
> and what I am looking for is UHJ stereo to a speaker feed. I really do not
> need the ability to adjust to different 4 channel layouts like the Onkyo
> provided and could just live with a square.
>
>
> I could be I simply missed something in the ATK as maybe a signal chain
> that could provide the decoding I need to convert my old CD's and LP'sbut
> thought why not see if the knowledge base here might be aware of a solution
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Re: [Sursound] Nimbus promo video

2021-12-13 Thread Richard G Elen


Well, that's rather a neat find. Never saw that before!
Richard E

On 13/12/2021 19:45, Eero Aro wrote:

Hi All

I uploaded a Nimbus Records Ambisonics promotion video into YouTube:

https://www.youtube.com/watch?v=DrGJxlrv08M

All audio in the video is UHJ encoded Ambisonics. The video includes 
presentation
about Ambisonics and several video/audio clips, for example from a 
concert, Zoo,

steam railway and from the Farnborough Air Show.

Eero
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Re: [Sursound] Playback systems for surround installation

2021-05-17 Thread Richard Foss

Hi Ezra,

You could look at our ImmerGo system as a possibility for the performance and 
the later playback.

For the performance you could use a MOTU 24Ao or Presonus StudioLive 64S/32S 
and move USB, analog or ADAT input channels in 3D space. If you have a 
different mixing console with an accessible control protocol, we could update 
ImmerGo to work with it.

For the later playback, you could create a multichannel wav file from the 
recording and play that back in a loop. There is a scheduler that will allow 
for the looping of one or more multichannel wav files.

Our website has further information.

Richard.

---
Richard Foss (PhD)
Software engineer/director

ImmersiveDSP
46 Silver Ranch Estate
Keurbooms River Road
Plettenberg Bay 6600
South Africa

Email: rich...@immersivedsp.com
Web:  www.immersivedsp.com
Cell:   +27832889354

> On 16 May 2021, at 17:40, Cape Wrath  wrote:
> 
> Hello,
> We will be creating our first surround sound/music performance for a
> festival.
> 
> After the performance our piece will be set up as a 12 speaker standalone
> installation playing back on a loop.
> 
> Could you please recommend a playback system/setup?
> We’d like to bounce the piece down to one surround file which can then be
> played back through the system. What would be the best format to bounce the
> session down to?
> 
> Thank you in advance,
> Ezra.
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Re: [Sursound] DBADP

2020-11-17 Thread Richard Foss

Hi Dave,

Getting to your questions:

> By “current products” I meant hardware audio mixers and DSP processors with 
> multiple inputs and outputs that are readily available, rather than what can 
> be built using things like the Sharc DSP chips or cards. This is certainly 
> beyond my capabilities, but seems to be what you and others are doing.

Yes, I did understand your meaning, and MOTU products were the first of the 
“current products” we have used. We are moving onto others now.

Regarding the miniDSP speaker implementation, you are correct, the volume/delay 
matrix was specially built for the Sharc chip inside the PoE AVB speakers.

Your description of Spatial Audio processors matches my understanding. The 
Dolby Atmos spatial algorithm is an interesting one - from what I recall they 
have a 'dual balance' or even 'triple balance' panner for their object based 
channels, particularly good for movie theatres.

> I presume that each of your speakers receives the 32 output channels from the 
> matrix

A 32 x 2 matrix sits inside each speaker, not in the computer. The 2 output 
channels are there because each speaker has an amplifier for a further 
ancillary speaker. Filtering happens after the matrix. We send the same 32 
channels to each speaker. The matrix cross point values for volume and delay 
are controlled via AVB control messages from the computer. We do have to smooth 
the delay changes. As I mentioned, this allows  system processing capability to 
grow incrementally with speaker additions.

> I’ve looked again at the available MOTU products, which are excellent audio 
> interfaces, but they alone do not seem to provide what is needed for an SAE.

The MOTU devices, like many mixers have multiple aux buses, each of which can 
be assigned to an analog output for speaker connection. Each input channel can 
be assigned to an analog/USM/ADAT/.. input. Each aux bus can have a 
controllable mix of input channels. So you effectively have a mixer matrix 
inside the device. The control is via OSC commands over Ethernet. Updates are 
driven by quarter frame messages derived from a DAW, or from our own 
multichannel wav player, depending on where you want your sound sources to come 
from. As I mentioned, our goal was to move a substantial part of the spatial 
processing to currently available devices.

I hope that answers your questions.

Regards,

Richard.

---
Richard Foss (PhD)
Software engineer/director

ImmersiveDSP
46 Silver Ranch Estate
Keurbooms River Road
Plettenberg Bay 6600
South Africa

Email: rich...@immersivedsp.com
Web:  www.immersivedsp.com
Cell:   +27832889354

> On 17 Nov 2020, at 14:39, Dave Hunt  wrote:
> 
> Hi Richard,
> 
> In the interests of avoiding too much repetition, I’ve edited the previous 
> conversations.
> 
> By “current products” I meant hardware audio mixers and DSP processors with 
> multiple inputs and outputs that are readily available, rather than what can 
> be built using things like the Sharc DSP chips or cards. This is certainly 
> beyond my capabilities, but seems to be what you and others are doing.
> 
> Having looked at your website, I’ve been trying to understand your approach 
> to this, and how it may agree with or differ from mine and others'. 
> 
> A large number of inputs needs to be mixed to a large number of outputs in a 
> controlled way. A 32 channel in/out matrix mixer with independent amplitude, 
> delay (and possibly other processing) at the crosspoints for each input to 
> each output is indeed desirable. The inputs are mixed to the outputs in a way 
> that depends on the spatial algorithm (ambisonics, Dolby Atmos, VBAP, DBAP, 
> WFS etc.). This requires computer control as the number of instructions is 
> large. If an input sound source needs to move spatially the instructions need 
> to be smoothed, to avoid it jumping between positions, with a ramp or curve 
> driven by time.
> 
> All this could be built into a digital mixer, but incorporating a good user 
> interface is far from trivial, and this is unlikely to happen as general 
> demand for it is low, and it will be expensive.
> 
> So, we end up with a separate DSP spatial audio engine (SAE for short) that 
> sits between a large mixer or DAW sending many channels, and a large number 
> of amplifiers and speakers. The connections are best done with a digital 
> audio network (AVB, Dante, MADI or other). The mixer and DAW are used 
> relatively normally, though mixes are to several “stems”, which are then 
> "spatialised" , rather than to a stereo or 5.1 output. This avoids adding 
> extra processing loads to the mixer or DAW. This SAE could be software in 
> another, or even the same computer, though processing load and latency would 
> be problematic. Less so when using a DAW (even with video) than with realtime 
> events.
>

Re: [Sursound] DBADP

2020-11-15 Thread Richard Foss


Thanks for the delta stereophony history Dave, interesting!

> Current products do not allow progress to true Delta Stereophony (DBADP)


Well conceptually it should be possible if, beyond aux mixes, you have a 
further layer of mixes that can comprise aux bus sends (with controllable 
delays/filtering/volumes) as well as input channels. A possible problem is not 
having sufficiently small delay increments, and not having smoothing within the 
device. Anyway, its worth doing some experimentation! Implementing DBAP or VBAP 
is fine.

> DSP chips are now capable of providing it


Yes, there is a Sharc DSP in the miniDSP speakers we use, and a controllable 
32x2 matrix with delays/attenuation at the cross points.

As you say, running Spat and a DAW is processor intensive. This was one of the 
reasons we have turned to using the processors in current devices to do the 
post-render mixing/delays. Having this capability in a speaker is great, 
because your processing capability grows with each speaker. Having it in an 
audio interface/mixing desk means that all the inputs - analog/usb/ADAT/… can 
have spatialisation applied to them.

> On 15 Nov 2020, at 13:56, Dave Hunt  wrote:
> 
> Hi Richard,
> 
> I’ve changed the title of this topic to something more relevant. 
> 
> I still prefer the term Delta Stereophony to describe this. It seems to date 
> back to the mid 1980’s, and was described by Gerhard Steinke and Wolfgang 
> Ahnert. They were working in East Germany behind the Iron Curtain, reputedly 
> working with Sinclair ZX Spectrum computers and expensive AKG delay lines 
> somehow imported from Austria.
> 
> It does make a great deal of sense. When digital delay lines became more 
> generally available and affordable (1990’s ???) they were increasingly used 
> in public address systems to improve coverage over a greater area, using 
> speakers down the length of an auditorium to augment the usual left/right or 
> LCR main frontal system. The feed to these was delayed by an amount that 
> caused the time of arrival of sound from them to match that of the main 
> frontal system . Sometimes the feed to a "front fill” system, arrayed along 
> the front of the stage to increase clarity in the rows of seating near the 
> stage, was also delayed to match the time of arrival of sound from its 
> source. Amplitudes were usually adjusted by ear, as indeed were delay times 
> after an initial calculation.
> 
> These systems were more “appropriately distributed mono” than spatial. It is 
> impossible to get the delay/amplitude combination correct for every position 
> in the space with a finite number of speakers and output channels, so 
> compromises are inevitable. This became common practice, especially for large 
> scale stadium events. Digital mixing desks now commonly incorporate delays on 
> each output, making this simpler to implement.
> 
> Current products do not allow progress to true Delta Stereophony (DBADP), as 
> the architecture does not provide delay as well as amplitude control on each 
> matrix crosspoint, and the market doesn’t expect or demand it. DSP chips are 
> now capable of providing it, as proved by TiMax, LISA, d’s Soundscape, 
> Iosono , Astro, and Meyer’s relaunched system. The market is small, and the 
> DSP boxes pricey. It becomes relatively more affordable for large 
> multi-speaker systems with large budgets.
> 
> For the rest of us, it’s down to software. Ircam have a basic implementation 
> of DBAP in Spat~ for Max/MSP (or you can roll your own), and adding the delay 
> component is relatively simple. You can then scale the amplitude and delay 
> separately for each source, as seems appropriate. Using delay alone is 
> surprisingly effective. The variation of amplitude between widely spaced 
> speakers can be excessive.
> 
> Of course you need a fast and powerful computer, and efficient programming to 
> do this, but that is also true with any of the alternative algorithms 
> (ambisonics, VBAP, DBAP, WFS etc.). None of these are perfect for every 
> situation, and it is hard to envisage a combination of them that would work.
> 
> Ciao,
> 
> Dave Hunt
> 
> 
>> On 14 Nov 2020, at 17:00, sursound-requ...@music.vt.edu wrote:
>> 
>> From: Richard Foss 
>> Subject: Re: [Sursound] Was: Recorder for ORTF-3D OUTDOOR SET
>> Date: 14 November 2020 at 16:48:36 GMT
>> To: sursound@music.vt.edu
>> 
>> 
>> Dave, I have meant to follow up on your message  for some time, because your 
>> ideas match what I am currently busy with - at last getting to it!
>> 
>> Our first immersive audio implementation uses networked PoE miniDSP speakers 
>> which each incorporate a matrix mixer with volume and delay control at the 
>> cross points.

Re: [Sursound] Was: Recorder for ORTF-3D OUTDOOR SET

2020-11-14 Thread Richard Foss

Dave, I have meant to follow up on your message  for some time, because 
your ideas match what I am currently busy with - at last getting to it!


Our first immersive audio implementation uses networked PoE miniDSP 
speakers which each incorporate a matrix mixer with volume and delay 
control at the cross points. The delays were a later addition, and I 
certainly found that the localization was enhanced by incorporating 
delays. We implemented DBAP for the amplitude panning, but we have 
implemented and experimented with VBAP. Given that our targeted 
applications will need irregular speaker configurations, we have settled 
on DBAP for now.


We had an idea, similar to yours, to utilize the signal processing 
capabilities of audio interfaces/mixers. Because we owned MOTU devices, 
we tried this first on three of the MOTU devices, and have updated our 
ImmerGo software to work with these interfaces. However, it was not 
possible to implement a delay matrix on the MOTU devices, so they just 
have a DBAP implementation, not DBADP (your innovative label:)).


I am now working on a mixing console implementation where I believe I 
can have delay/EQ at matrix cross points for a few channels, where there 
is an inverse relationship between number of speakers and number of 
channels with delay/EQ, although all channels can have DBAP. One does 
need to have mix buses to enable this, and also there often is the 
timing constraint, because a lot of messages go to the mixer as the 
sound sources are spatialized. I have found that the MOTU devices are 
very responsive in this regard.


Anyway, good to have a fellow DBADP enthusiast .

Regards,

Richard.

—
Richard Foss (PhD)
Software engineer/director

ImmersiveDSP
46 Silver Ranch Estate
Keurbooms River Road
Plettenberg Bay 6600
South Africa

Email: rich...@immersivedsp.com
Web:  www.immersivedsp.com
Cell:   +27832889354

On 2020/10/26 12:43 pm, Dave Hunt wrote:

Hi,

This conversation seems to have branched into a different topic: the 
effectiveness of various immersive audio methods in different circumstances.

Ambisonics and VBAP are suitable for regular, or near regular, loudspeaker 
arrays with a central listening area. DBAP is more suitable for wide area 
irregular arrays, or even zones within the array, as is 
one-loudspeaker-per-source sound for static sources.

A little mentioned alternative here is Delta Stereophony, where the source sound is 
fed to each speaker with an appropriate gain and delay based on the distance of the 
source from each speaker. TiMax is based on this approach. I’m sure that d’s 
Soundscape, LISA, Iosono, Astro and several other systems aimed at large scale 
immersive audio are also based on this. It could be regarded as stripped down wave 
field synthesis, using a small fraction of the number of speakers required for true 
WFS.

It could be called DBADP.

Because of the number of calculations required, especially for moving sources, 
the trend here is for a dedicated hardware multi-channel spatial audio 
processor controlled by software, or messages from a digital mixing desk. A 
common format for these messages seems to be OSC.

https://www.lightsoundjournal.com/2020/03/05/is-immersive-audio-coming-of-age/

It would be feasible to incorporate this, or indeed any of these processes, 
into digital mixing desks. though providing a good user interface is 
problematic. As to plug-ins, that depends on the demand and the power of 
computers.

SARS Covid 2 is likely to slow progress on this dramatically.

Ciao,

Dave Hunt



On 25 Oct 2020, at 16:00, sursound-requ...@music.vt.edu wrote:

From: Augustine Leudar 
Subject: Re: [Sursound] Recorder for ORTF-3D OUTDOOR SET
Date: 25 October 2020 at 12:56:11 GMT
To: Surround Sound discussion group 


I think perhaps Stefan there is a language barrier or problem with
communication here ?

Your exact words were ""*It is important to see that every position is
panned to 2 speakers in *
*2D, and (usually) 3 speakers in 3D.*"
You didn't refer to mono - but I did refer to "one" channel being spread
over several speakers sounding crap (or blurring spatialisation) to which
you replied this was wrong because its how millions of records were made -
- of course I believe you know what were stereo is and were referring to
panning one sound source (eg a trumpet)  across two speakers as in
stereophonic panning -  "one soundsource" being different to "one channel"
- thus the confusion?
Yes I know what VBAP is - I literally said it uses triangles in my last
post and that I've used it for 15 years, VBAP is great - I do mix it with
ambisonics sometimes though.
As for upmixing stereo to 5.1 - I agree with you about leaving the front
left and right intact if you *HAVE* to do it - all I was saying is upmixing
stereo to 5.1 is never going to be as good as actually having a 5.1 mix to
start with all six discrete channels created from the beginning uising a
surround sound p

Re: [Sursound] Funding opportunity for small business

2020-10-28 Thread Richard Dobson

Sounds like you need a web-based clone of Sonic Visualiser 
(https://www.sonicvisualiser.org/).


I see the grant is limited to USA small businesses.

Richard Dobson

On 28/10/2020 00:47, Anne Simonis wrote:

  Hi All,

I'm an acoustic ecologist with NOAA, and my research uses underwater
acoustic recordings to study marine mammal ecology, behavior, and the
impacts of human noise.

NOAA recently announced a small business grant opportunity, with a subtopic
related to citizen science and education (subtopic 9.5 in the full
announcement):
https://www.grants.gov/web/grants/view-opportunity.html?oppId=329444


...
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[Sursound] The birth of Ambisonics & the Soundfield Mike

2020-05-23 Thread Richard Lee

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

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

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

The Mk4 was, IM not so HO, the best microphone of the 20th Century.
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Re: [Sursound] analog planner

2020-02-17 Thread Richard Elen

The Audio & Design Recording boxes that Geoff Barton designed were 
intended as (analogue) outboard units that could be patched into a 
conventional mixing console to generate (mainly) first-order (all there 
was) Ambisonic B-Format.


The Pan-Rotate unit included eight 360-degree controls, each with a 
radius vector (distance from centre) control with a switched position 
for maximum radius vector. In addition there was a B-Format Converter 
which allowed console panpots to be used to pan across a quadrant, using 
four groups and an aux send.


Finally, a Transcoder unit generating 2-channel UHJ only could be fed 
with stereo signals for front and rear stages with a stage width control 
on each.


The units are described in these articles:

http://ambisonic.net/branwell_arb.html

http://ambisonic.net/ambimix.html

I'm afraid I don't have circuit diagrams but Geoff presumably has.

--R

On 16-Feb-20 15:18, Peter Carbines wrote:

On 16/02/2020 12:19, Augustine Leudar wrote:

I'm mainly interested in purely analog devices in this thread


The small team I was involved in during the 1970s built a 4-input 
proof-of-concept Ambisonic mixer. We were interested in developing a 
mixer to produce B-format from multitrack recordings, creating an 
artificial soundfield.


It had 360-degree panning ability and radius-vector control on each. 
Two input channels used sine-cosine 360 degree potentiometers derived 
from scrapped electronics, probably radar consoles. Because we were 
unfunded and basically running on 'pocket money', the budget didn't 
allow for 4 sine-cosine pots so we designed and built two switchable 
pan-pots, initially 30-degree but planned for an eventual 15 degrees. ...

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Re: [Sursound] FOA/HOA Impulse Responses

2019-10-10 Thread Richard Dobson

For fun, you can try the "Darkside" responses recorded a few years ago 
by illustrious members of this list:


http://www.rwdobson.com/sspaces/sciencespaces.html

Richard Dobson


On 09/10/2019 17:24, Mads Kjeldgaard wrote:

Hello everyone

I was wondering if there were some freely available impulse response 
recordings online in FOA or HOA format that could be used for ambisonic 
convolution reverb experiments, before I head out and record my own?


Best regards

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Re: [Sursound] Pan Law Irregular Speaker Array

2019-08-26 Thread Richard Foss

Hi Jonathan,

Thanks for pointing this out - yes, it a delay for each source at each speaker. 
This might have been what was confusing Ralph.

Maybe some context would help. Control is from a mobile device, which sends 3D 
coords to a Mac/PC server. A DAW or multi-channel Wav file plays 32 channels of 
audio over Ethernet AVB. These 32 channels get sent to each speaker. In each 
speaker is a 32 x 1 matrix with attenuation and delay at each cross point. I 
figure out the mix levels, using DBAP, and the delay levels using the virtual 
source to speaker distances.

I haven’t needed fast moving sounds, but would be great to have frequency 
manipulation at the cross points, and experiment with a Doppler effect. 

Hope that clarifies :)

> On 26 Aug 2019, at 18:36, Jonathan Kawchuk  wrote:
> 
> Hi Richard,
> 
> Very interesting. So there would be a delay consideration per source, not per 
> speaker. Is this already written into the DBAP algorithm or any pan law at 
> that, or do you manually implement. If so, how?
> 
> By the way, in much the same line of thinking, if this gunshot were moving, 
> would you implement a doppler effect or would that be impossible to calculate 
> on a per source basis since it is so listener position dependant?
> On Aug 23, 2019, 2:27 PM -0600, Richard Foss , wrote:
>> Sorry, slow reply Ralph. The way I see it - focus on the virtual source,
>> not the listener position. Wherever the virtual source is positioned in
>> an installation, make the speakers respond as if there was a real source
>> at the virtual source position. If there was a gunshot at the virtual
>> source position, the gunshot should not play immediately from a speaker
>> some distance away from the virtual source position.
>> 
>> A person close to the further speaker (from the virtual source) should
>> hear the louder gunshot from the close speaker (to the virtual source)
>> and the softer gunshot from the further speaker at the same time. If the
>> softer gunshot arrived from the further speaker first, the proximity
>> effect might kick in.
>> 
>> I am talking about a DBAP context, where all speakers play at varying
>> levels, not for example VBAP. As I mentioned, this approach seems to
>> work very well when implemented.
>> 
>> On 2019/08/22 10:18 PM, Ralph Jones wrote:
>>> Richard Foss, I still don’t get it, sorry. Perhaps I’m being obtuse. But to 
>>> clarify, you said:
>>> 
>>>>>> for a particular real source
>>>>>> channel, delay its play out from a speaker FAR from the virtual source
>>>>>> LONGER than from a speaker CLOSE to the virtual source
>>> 
>>> (capitalization mine). Why do you want to delay the signal to the FARTHER 
>>> speaker? How does that help address proximity effect? It seems to me that 
>>> it would only accentuate it.
>>> 
>>> Ralph Jones
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>> Professor Richard Foss
>> Computer Science Department
>> Rhodes University
>> Grahamstown 6140
>> South Africa
>> 
>> Tel: +27 46 6038294
>> Cell: +27 83 288 9354
>> email: r.f...@ru.ac.za
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Re: [Sursound] Pan Law Irregular Speaker Array

2019-08-23 Thread Richard Foss

Sorry, slow reply Ralph. The way I see it - focus on the virtual source, 
not the listener position. Wherever the virtual source is positioned in 
an installation, make the speakers respond as if there was a real source 
at the virtual source position. If there was a gunshot at the virtual 
source position, the gunshot should not play immediately from a speaker 
some distance away from the virtual source position.


A person close to the further speaker (from the virtual source) should 
hear the louder gunshot from the close speaker (to the virtual source) 
and the softer gunshot from the further speaker at the same time. If the 
softer gunshot arrived from the further speaker first, the proximity 
effect might kick in.


I am talking about a DBAP context, where all speakers play at varying 
levels, not for example VBAP. As I mentioned, this approach seems to 
work very well when implemented.


On 2019/08/22 10:18 PM, Ralph Jones wrote:

Richard Foss, I still don’t get it, sorry. Perhaps I’m being obtuse. But to 
clarify, you said:


for a particular real source
channel, delay its play out from a speaker FAR from the virtual source
LONGER than from a speaker CLOSE to the virtual source


(capitalization mine). Why do you want to delay the signal to the FARTHER 
speaker? How does that help address proximity effect? It seems to me that it 
would only accentuate it.

Ralph Jones
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Re: [Sursound] Pan Law Irregular Speaker Array

2019-08-20 Thread Richard Foss

Ralph, I'm thinking of an installation context, where there are many 
listening positions. If a person is standing close to a speaker that is 
some distance from the position of a virtual sound source, you would 
want an appropriate delay in the playback of the sound source from the 
close by speaker. In DBAP it will be at a lower amplitude than the 
speakers close to the virtual source, but the precedence effect could 
well make it appear that the source is at the close by speaker.


For our immerGo immersive sound system I made use of attenuation and 
delays at the cross points of a matrix in each of a number of speakers. 
I found that the incorporation of delays, together with attenuation 
values generated from DBAP, contributed substantially to the 
localization effect.


On 2019/08/20 9:15 PM, Ralph Jones wrote:

I believe this is backwards. For a given listening position, the *closer* 
speaker should be delayed if you want its output to arrive at the same time as 
that from a farther speaker.

Richard Foss wrote:


Just catching up on your comment about time alignment with DBAP, Gus.

I do think its useful to use delays - so for a particular real source
channel, delay its play out from a speaker far from the virtual source
longer than from a speaker close to the virtual source. With DBAP the
source channel will play from all speakers, albeit with different
amplitudes. For a person in an installation standing close to a further
speaker, the precedence effect could well come into play without
appropriate delays, as I see it.


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Re: [Sursound] Pan Law Irregular Speaker Array

2019-08-19 Thread Richard Foss

Just catching up on your comment about time alignment with DBAP, Gus.

I do think its useful to use delays - so for a particular real source
channel, delay its play out from a speaker far from the virtual source
longer than from a speaker close to the virtual source. With DBAP the
source channel will play from all speakers, albeit with different
amplitudes. For a person in an installation standing close to a further
speaker, the precedence effect could well come into play without
appropriate delays, as I see it.

On 2019/08/18 1:26 PM, Augustine Leudar wrote:

also you could use rew to eq each individual speaker. There's not much
point time aligning DBAP because time align to where ? The beuaty of using
amplitude panning with irregualr speaker arrays is that there's no sweet
spot, especially for walk through installations.

On Sun, 18 Aug 2019 at 12:23, Augustine Leudar
wrote:

*should say "the software doesn't know you've placed the speakers in a
line not a circle"
Its a crude work around - but it works if you dont have time to learn
ICST. You can also try jamona.

On Sun, 18 Aug 2019 at 12:21, Augustine Leudar
wrote:

I use ICST in MAx MSP for irregular speaker arrays and it works fine even
though it says its "Ambisonics equivelant panning" if you reduce the
directivity it basically works like a form of DBAP. It works really well,
Ive made sound sfly through mazes and up and down buildings with it. You
can even use a normal octaphonic panner to make things fly in a straight
line - at the end of the day the sound is traveling from one speaker to the
next - the software doesnt know youve placed the software in a line rather
than a circle.

On Sat, 17 Aug 2019 at 23:33, Jonathan Kawchuk <
jonathan.kawc...@gmail.com> wrote:

Hey everyone,

I’m having a bit of trouble setting up a SPAT Revolution session for an
irregular speaker array.

1. I am working with an irregular array of speakers for an installation
(see attached, speakers numbered 1-12). The speakers are in a long hallway
and are irregularly placed in the x, y, and z planed though always
overhead. I would like to have sound move with each walker as they pass
through the hallway by automating sound to move at average human walking
speed. I am using Speaker 7 as an origin point (0, 0, 0).

A. Which is the correct pan law to use? I have tried DBAP and KNN but
the diagram of each speaker seems to be pointing in the wrong direction,
esp. at the origin. I would like the speakers to point down towards the
floor.

B. Is it advised to select a single speaker as the origin point?

2. I have taken surround impulse responses from an array of 7 speakers
in a surround setup approx 19m from the origin. The microphone array was
7+4 (height) at 1m from the origin. If spatializing in SPAT, would I make
the convolved signal 19m from the origin (where the speakers played the
sweep) or 1m (where the microphone array was).

3. In either case, would I compute speaker alignment or normalize?

4. Are you aware of any products that can calculate multichannel
irregular speaker distances/correction curves from a test signal? Something
like sonarworks or genelec SAM but not proprietary and with the ability to
calculate actual speaker positions from test signals rather than having to
scan the room (which I currently do using matterport) or measure by hand.

Thanks so much,

Jon

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Re: [Sursound] Ambisonic Hand Controller

2019-08-03 Thread Richard Foss

Hi Marc,

I use the node.js socket.io <http://socket.io/> module for the transmission of 
the 3D data to the server - its a library that uses WebSockets.

Richard.


> On 02 Aug 2019, at 19:13, Marc Lavallée  wrote:
> 
> Hi Richard,
> 
> My question was not directly related to the orientation API, but to the 
> method used to transmit its data to your server. You mentioned https;  either 
> https AJAX or secure WebSockets (WSS) can be initiated by a https request. 
> Maybe a realtime connection using WebSockets is not required, slow AJAX 
> requests being fast enough. I was just curious.
> 
> Marc
> 
> 
> Le 19-08-02 à 12 h 31, Richard Foss a écrit :
>> Hi Marc,
>> 
>> The following will give you a better idea of the Web API capabilities that I 
>> am using:
>> 
>> https://developer.mozilla.org/en-US/docs/Web/API/Detecting_device_orientation
>>  
>> 
>> Richard.
>> 
>> 
>> On 2019/08/02 4:38 PM, Marc Lavallée wrote:
>>> Hi Richard.
>>> 
>>> Without reading access to the (paywalled) article and (unpublished) code, 
>>> it's difficult to understand all the effects of this problem. So I 
>>> wondered: are the events send via AJAX or WebSockets?
>>> 
>>> Marc
>>> 
>>> Le 19-08-02 à 09 h 53, Richard Foss a écrit :
>>>> Bo-Erik, we have always been able to open a connection  to the web server 
>>>> and upload code. The problem is that access to the mobile device's device 
>>>> orientation events by our code on the client is disallowed when using 
>>>> chrome and an http server - fine with other web browsers. This is a recent 
>>>> phenomenon. When we make our web server secure (https), we can access the 
>>>> events. This requires the secure certification process.
>>>> 
>>>> On 2019/08/02 1:17 PM, Bo-Erik Sandholm wrote:
>>>>> As long as the hosting server is a local host websocket server chrome can
>>>>> open a connection to the server.
>>>>> At least the version available a month ago.
>>>>> 
>>>>> Den fre 2 aug. 2019 10:28Richard Foss  skrev:
>>>>> 
>>>>>> Hi Justin,
>>>>>> 
>>>>>> I have described the use of a mobile device controller in the following
>>>>>> paper that is in the AES e-library:
>>>>>> 
>>>>>> http://www.aes.org/e-lib/browse.cfm?elib=19714
>>>>>> 
>>>>>> Its a client server system, where the server is a node.js server and a
>>>>>> browser is used to run client code. This means I can use the Web API
>>>>>> device orientation capabilities. One frustrating issue is that Chrome
>>>>>> has recently stopped allowing device orientation access unless the
>>>>>> server is secure.
>>>>>> 
>>>>>> Richard Foss.
>>>>>> 
>>>>>> On 2019/07/26 9:01 PM, Justin Kuhn wrote:
>>>>>>> Hi Everyone,
>>>>>>> 
>>>>>>> I am looking into making a controller in which I harness roll, pitch, 
>>>>>>> and
>>>>>>> yaw from my phone or a vr controller for real-time control of the
>>>>>>> orientation of a soundfield.  I want to hook it up using OSC to Reaper 
>>>>>>> or
>>>>>>> Supercollider either wirelessly (preferred) or through USB (if I have
>>>>>> to).
>>>>>>> I have a few reasons I want to do this.
>>>>>>> 
>>>>>>>- I want to do real-time encoding and decoding with a controller that
>>>>>> can
>>>>>>> add expressive qualities with my hands for 4.0 performance purposes
>>>>>>>- I want to better calibrate binaural playback through traditional
>>>>>>> headphones for production purposes
>>>>>>>- I hope to avoid using a headset for ambisonic production entirely.
>>>>>> It's
>>>>>>> so much easier to turn my hand around very quickly than it is to use my
>>>>>>> head.
>>>>>>>- Would make a cool pairing with Wekinator.
>>>>>>> 
>>>>>>> Has anyone done this, or seen it done?  I'd love some tips and
>>>>>> suggestions
>>>>>>> before I get started.
>>>>>>> 
>>>>>>> Thanks!
>>>>>>&g

Re: [Sursound] Ambisonic Hand Controller

2019-08-02 Thread Richard Foss

Hi Marc,

The following will give you a better idea of the Web API capabilities
that I am using:

https://developer.mozilla.org/en-US/docs/Web/API/Detecting_device_orientation

Richard.

On 2019/08/02 4:38 PM, Marc Lavallée wrote:

Hi Richard.

Without reading access to the (paywalled) article and (unpublished)
code, it's difficult to understand all the effects of this problem. So
I wondered: are the events send via AJAX or WebSockets?

Marc

Le 19-08-02 à 09 h 53, Richard Foss a écrit :
Bo-Erik, we have always been able to open a connection to the web
server and upload code. The problem is that access to the mobile
device's device orientation events by our code on the client is
disallowed when using chrome and an http server - fine with other web
browsers. This is a recent phenomenon. When we make our web server
secure (https), we can access the events. This requires the secure
certification process.

On 2019/08/02 1:17 PM, Bo-Erik Sandholm wrote:
As long as the hosting server is a local host websocket server
chrome can

open a connection to the server.
At least the version available a month ago.

Den fre 2 aug. 2019 10:28Richard Foss skrev:

Hi Justin,

I have described the use of a mobile device controller in the
following

paper that is in the AES e-library:

http://www.aes.org/e-lib/browse.cfm?elib=19714

Its a client server system, where the server is a node.js server and a
browser is used to run client code. This means I can use the Web API
device orientation capabilities. One frustrating issue is that Chrome
has recently stopped allowing device orientation access unless the
server is secure.

Richard Foss.

On 2019/07/26 9:01 PM, Justin Kuhn wrote:

Hi Everyone,

I am looking into making a controller in which I harness roll,
pitch, and

yaw from my phone or a vr controller for real-time control of the
orientation of a soundfield. I want to hook it up using OSC to
Reaper or

Supercollider either wirelessly (preferred) or through USB (if I have

to).

I have a few reasons I want to do this.

- I want to do real-time encoding and decoding with a
controller that

can

add expressive qualities with my hands for 4.0 performance purposes
- I want to better calibrate binaural playback through traditional
headphones for production purposes
- I hope to avoid using a headset for ambisonic production
entirely.

It's
so much easier to turn my hand around very quickly than it is to
use my

head.
- Would make a cool pairing with Wekinator.

Has anyone done this, or seen it done? I'd love some tips and

suggestions

before I get started.

Thanks!

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Re: [Sursound] Ambisonic Hand Controller

2019-08-02 Thread Richard Foss

Bo-Erik, we have always been able to open a connection to the web
server and upload code. The problem is that access to the mobile
device's device orientation events by our code on the client is
disallowed when using chrome and an http server - fine with other web
browsers. This is a recent phenomenon. When we make our web server
secure (https), we can access the events. This requires the secure
certification process.

On 2019/08/02 1:17 PM, Bo-Erik Sandholm wrote:

As long as the hosting server is a local host websocket server chrome can
open a connection to the server.
At least the version available a month ago.

Den fre 2 aug. 2019 10:28Richard Foss skrev:

Hi Justin,

I have described the use of a mobile device controller in the following
paper that is in the AES e-library:

http://www.aes.org/e-lib/browse.cfm?elib=19714

Richard Foss.

On 2019/07/26 9:01 PM, Justin Kuhn wrote:

Hi Everyone,

I am looking into making a controller in which I harness roll, pitch, and
yaw from my phone or a vr controller for real-time control of the
orientation of a soundfield. I want to hook it up using OSC to Reaper or
Supercollider either wirelessly (preferred) or through USB (if I have

to).

I have a few reasons I want to do this.

- I want to do real-time encoding and decoding with a controller that

can

It's

so much easier to turn my hand around very quickly than it is to use my
head.
- Would make a cool pairing with Wekinator.

Has anyone done this, or seen it done? I'd love some tips and

suggestions

before I get started.

Thanks!

*Justin Kuhn*
360/VR Video and Audio Production
justinkuhnmedia.com
Cell | 828-403-3171
He, Him, His
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Re: [Sursound] Ambisonic Hand Controller

2019-08-02 Thread Richard Foss

Hi Justin,

I have described the use of a mobile device controller in the following
paper that is in the AES e-library:

http://www.aes.org/e-lib/browse.cfm?elib=19714

Richard Foss.

On 2019/07/26 9:01 PM, Justin Kuhn wrote:

Hi Everyone,

- I want to do real-time encoding and decoding with a controller that can
add expressive qualities with my hands for 4.0 performance purposes
- I want to better calibrate binaural playback through traditional
headphones for production purposes
- I hope to avoid using a headset for ambisonic production entirely. It's
so much easier to turn my hand around very quickly than it is to use my
head.
- Would make a cool pairing with Wekinator.

Has anyone done this, or seen it done? I'd love some tips and suggestions
before I get started.

Thanks!

*Justin Kuhn*
360/VR Video and Audio Production
justinkuhnmedia.com
Cell | 828-403-3171
He, Him, His
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Tel: +27 46 6038294
Cell: +27 83 288 9354
email: r.f...@ru.ac.za

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Re: [Sursound] Interatcive visuals with interaactive spatial audio

2019-07-10 Thread Richard Foss

Thanks Gus, looks great! How are the movements being tracked?

Richard.


Professor Richard Foss
Computer Science Department
Rhodes University
Grahamstown 6140
South Africa

Tel: +27 46 622 8294
Cell: +27 83 288 9354



> On 10 Jul 2019, at 16:12, Augustine Leudar  wrote:
> 
> Should of course say  Interactive visuals with Interactive spatial audio!!
> 
> On Wed, 10 Jul 2019 at 15:09, Augustine Leudar  <mailto:augustineleu...@gmail.com>>
> wrote:
> 
>> I thought this might be of interest to some of you. I have been working
>> with a visual arts studio recently who create interactive visuals that
>> track peoples movements and convert them into strange shapes on the screen.
>> At the moment its on large screens so I have speakers lining the edge and
>> behind the screen so that the audio tracks the silhouettes as they cross
>> the screen though I look forward to working with dome projections (contact
>> me if you do dome projections !!) . I use granular synthesis to morph
>> sounds in response to dta and spatialise with max so theres a really nice
>> integration between the movement of the people and the sounds and visuals.
>> This project was on a very large screen at the Athens science festival this
>> year :
>> 
>> https://vimeo.com/341857013
>> 
>> all the best
>> Gus
>> --
>> Dr. Augustine Leudar
>> Artistic Director Magik Door LTD
>> Company Number : NI635217
>> Registered 63 Ballycoan rd,
>> Belfast BT88LL
>> www.magikdoor.net
>> +44(0)7555784775
>> 
>> 
> 
> -- 
> Dr. Augustine Leudar
> Artistic Director Magik Door LTD
> Company Number : NI635217
> Registered 63 Ballycoan rd,
> Belfast BT88LL
> www.magikdoor.net <http://www.magikdoor.net/>
> +44(0)7555784775
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[Sursound] practical HOA encoding

2019-06-11 Thread Richard Lee

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

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

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

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

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

MY SOLUTION TO LOADSA LF GAIN FOR HOA ENCODING

Ignore it.
___

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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


Back to HOA encoding.

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

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

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

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

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

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

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

So in most cases, just use LF boost on

Re: [Sursound] Deconstructing soundbar marketing B.S.

2019-05-28 Thread Richard Dobson





On 28/05/2019 08:49, Augustine Leudar wrote:

wow :


.. It would be great if someone could

just invent bluetooth quad (or more)  then you could happily just deposit
four cable free bluetooth speakers around the room and be done with all
this nonsense.


...
Slightly a propos to this: in his 2000 "Millennium" article in JAES, 
Andy Moorer proposed "In 20 years, loudspeakers and microphones will 
know where they are". How are we doing with that, with one year to go?


Richard Dobson
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Re: [Sursound] Enquiry on upmixing from 1st order ambisonics to 3rd order ambisonics.

2019-02-22 Thread richard ford

Interesting question David.

  At 02:37 22-02-19, Wilson Lim wrote:

>ot sure if I have missed a discussion about upmixing with ambisonics on
>Sursound.
>
>Just wondering if anyone is willing to share some information on how to
>implement upmixing algorithms from 1st Order Ambisonics A to 3rd
>Order Ambisonics B-format.

I am curious to know what advantage there is to playing back 1st 
order upmixed to 3rd order. Doesnt it still sound like 1st order, 
since there no information is actually added? Or are the images 
expectated to be more stable on account of more loudspeakers being 
involved, on the analogy of stereo played back through three loudpeakers?

David

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Re: [Sursound] Hardest wearing outdoor speakers

2019-01-25 Thread Richard Lee

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

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

And what is an Extreme 5 ?
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Re: [Sursound] Passive small speaker advice

2019-01-13 Thread Richard Foss

Bruce, considering that price is an issue for you, you could look at the 
miniDSP SPK-4. It has an interesting cousin, the SPK-4P. The SPK-4P is an 
Ethernet AVB-capable speaker, and can get power, audio and control over the 
Ethernet cable. This eases configuration. The SPK-4 can also power an attached 
SPK4. You could have a configuration with up to 120 speakers.

Professor Richard Foss
Computer Science Department
Rhodes University
Grahamstown 6140
South Africa

Tel: +27 46 622 8294
Cell: +27 83 288 9354



> On 11 Jan 2019, at 1:35 PM, Bruce Wiggins  wrote:
> 
> Thanks to all for suggestions.  Really appreciated.  Definitely some
> possible solutions in there, depending on how much money I can wangle.
> 
> cheers
> 
> Bruce
> 
> 
> 
> On Fri, 11 Jan 2019 at 09:01, Axel Drioli  wrote:
> 
>> I second KEF egg E301. Coaxial, great sounding! Been using them for two
>> years now in any sort of situation, from studio to installations.
>> 
>> 
>> On Thu, 10 Jan 2019 at 23:54, Sönke Pelzer 
>> wrote:
>> 
>>> There aren't many loudspeakers in that test, but the measured Genelec
>> 8351A
>>> is the winner in nearly every category (if you take the time and look at
>>> the data, e.g. homogeneity and smoothness of directivity across frequency
>>> and angles, distortions, etc.)
>>> 
>>> Am Do., 10. Jan. 2019 um 02:24 Uhr schrieb Sven König :
>>> 
>>>> On 01/09/2019 02:27 PM, Bruce Wiggins wrote:
>>>>> Hello Sursounders (and Happy New Year!)
>>>>> 
>>>>> For many years, we've been running our Sounds in Space research
>>> symposium
>>>>> using lots of old passive minipod speakers (as Peter Lennox had lots
>> of
>>>>> these at the time of our first symposium, an we've been using them
>> ever
>>>>> since https://www.podspeakers.com/)
>>>> hi bruce,
>>>> 
>>>> kef ls50 were suggested before but if the budget is tight you should
>>>> also consider kef q150.
>>>> for ambisonics the kefs are actually the only speakers that are really
>>>> usable when you need a big sweetspot because of their wide and even
>>>> dispersion. i've never understood why in so many places adams, genelecs
>>>> or neumanns are used - their dispersion is really poor.
>>>> here's an interesting comparison:
>>>> https://www.princeton.edu/3D3A/Directivity.html
>>>> 
>>>> sven.
>>>> 
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>>> edit account or options, view archives and so on.
>>> 
>> --
>> *Axel Drioli*
>> *axeldrioli.com <http://axeldrioli.com/>*
>> 
>> *For immersive recordings visit ImmersiveFieldRecording.com
>> <http://immersivefieldrecording.com/>*
>> *For Spatial Audio production visit SpatialAudioLabs.com
>> <http://SpatialAudioLabs.com>*
>> 
>> *Tel-Facetime: +44 7460 223640*
>> *Skype: axel.drioli*
>> *E-mail: a...@spatialaudiolabs.com *
>> 
>> 
>> 
>> 
>> *Mostly LDN. UK.*
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Re: [Sursound] Looking for mic advice

2018-08-12 Thread richard ford

Are there any recordings available made using these 14mm electrets, and is a 
complete tree setup commercially available?
R

Sent from Yahoo Mail on Android 

  On Sun, 12 Aug 2018 at 11:21, Chris Woolf wrote:   
On 11/08/2018 18:16, jack reynolds wrote:
> I use 14mm electrets, so you can still get them pretty close together. They
> naturally have a lower noise floor and wide dynamic range.

Thanks for the clarification - I'd only seen your large diaphragm mics. 
14mm capsules sounds fine.

Chris Woolf
>
> Jack
>
> On 11 August 2018 at 14:42, Chris Woolf  wrote:
>
>> On 11/08/2018 10:59, Axel Drioli wrote:
>>
>>> ...  I use
>>> a prototype made by Reynolds Microphones. ... This mic has much lower
>>> self-noise than any other ambi mic you find around.
>>>
>>> But is that done using large diaphragm capsules? With the inevitable
>> consequences in terms of coincidence?
>>
>> Chris Woolf
>>
>>
>> ---
>> This email has been checked for viruses by Avast antivirus software.
>> https://www.avast.com/antivirus
>>
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>

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Re: [Sursound] Upcoming update to AmbiExplorer

2018-04-20 Thread richard ford

It's good to see the continut'growth in Ambisonics, but looking at your play 
store page for this app the general public needs to be made more aware of what 
Ambisonics is and its benefits fot the future of 'surround sound'
And on that point, sort of, i do have a question for Hector. You state on the 
Play Store that the app "decodes to UHJ". Now that sounds confusing, is it 
worded wrongly or is there something i'm missing here?
R

Sent from Yahoo Mail on Android 
 
  On Fri, 20 Apr 2018 at 5:08, Hector Centeno wrote:   Hello,

After a long while, I've decided it is finally time to update my Android
app AmbiExplorer [
https://play.google.com/store/apps/details?id=net.hcenteno.ambiexplorer ].
I'm developing it on my spare time but I'm hoping to include at least these
new features:

- Add HOA playback, up to 3rd order using Google's Resonance decoder (for
now, later will try to add a variety of HRTF selections)
- Include Google's Resonance decoder as an option for FOA playback in
addition to the current HRTF sets
- Support both FuMa and AmbiX formats
- Possibly support TBA (Facebook 360 Audio)
- Support loading custom HRTFs (I also want to look into possibly adding
SOFA support)
- I've also considered adding a VR mode where the user would be placed in a
visually neutral space (dimly lit) as an alternative to head-tracked
headphones (this might also elicit a more focused listening attitude), and
possibly add the option of assigning 360 panoramas to audio files to be
displayed during playback.

These updates will be released in stages during this year but the first 3
are already on their way and hopefully will be released soon.

If anyone has any feedback about these updates or would like to suggest any
other features or changes please let me know!

Best regards,

Hector Centeno
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Re: [Sursound] New book: Parametric time-frequency-domain spatial audio

2017-12-15 Thread Richard

This is basically aimed at those living in the UK.
I have contact with a well known artist who is performing an acoustic concert 
in the first quarter of 2018 that is being recorded for release on DVD. I have 
suggested that, considering where the event is taking place, that the audio be 
recorded Ambisonically and used as the source for the surround (possibly 4.0) 
track on the DVD as well as a UHJ encoded CD release.
For obvious reasons i can’t publicly give anymore information as things are 
still in motion and are a little hush-hush, so i’d be grateful if anyone knows 
of anyone who is capable of doing a professional job of recording Ambisonically 
here in the UK.
Fingers crossed
Richard
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Re: [Sursound] Simple Software to Play a 6-channel WAV File

2017-10-26 Thread Richard Dobson

Perhaps the "simplest" possible player is a command line one? My 
"paplay" program plays (and loops) multichannel files to a nominated 
device, and while it does not have interactive facilities it offers full 
channel mapping and selecting. Plus elementary B-Format decoding.


Part of the old mctoolkit, now resident here:

http://www.rwdobson.com/mctools.html

and also included in of the even older CDP system :-).

I am toying with doing a GUI counterpart, but not for a little while yet.

Richard Dobson

On 25/10/2017 23:14, len moskowitz wrote:

I wrote:

I'd appreciate it if someone could recommend a very simple Windows 
audio player that can play a 6-channel WAV file to a 6-speaker ring.


So far, the best solution is Wavosaur (https://www.wavosaur.com). The 
price is right: free, but donations are appreciated. I sent them 20 Euro.


It's a very light DAW program - just a single executable file - and 
doesn't need to be installed.


I copied it onto the Windows 10 PC, ran it, opened the 6-channel WAV 
file, set the audio card to ASIO (MME didn't work right), easily routed 
the 6 channels to the correct speakers for the hexagon speaker ring, and 
hit the space bar to play the file.


It was that simple.

It has one other nice feature that's useful for troubleshooting: you can 
mute any of the channels while it's playing by typing the number of the 
channel. Typing it again enables the channel again.


I'd like to have found an even simpler program, but this is likely as 
simple as it will get.



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




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Re: [Sursound] Multichannel players for permanent installations

2017-06-30 Thread Richard Foss

Completely blown away by the video! This is truly wonderful and an exemplary 
use of immersive sound.

The leap motion/Unity project was a post grad project. I will send a paper on 
it to you by email. Just need to check with conference organizers whether I can 
make it more publicly available.


> On 30 Jun 2017, at 11:29 AM, Augustine Leudar <augustineleu...@gmail.com> 
> wrote:
> 
> Hi Richard - Ive also being using the leap a lot - its great but a bit
> buggy sometimes when you get outside the infrared area , you can see it
> used in a big neolithic site here :
> 
> https://vimeo.com/217058501
> 
> I had a friend who used the connect which was also fine but a bit buggy.
> Do you have a link to the stuff you did with the leap ?
> anyway yes Ill message you near the time ,
> best,
> Gus
> 
> On 30 June 2017 at 10:23, Richard Foss <r.f...@ru.ac.za> wrote:
> 
>> That’s an inspiring video, thanks!
>> 
>> And many points of contact. A few years back we used the Microsoft Kinect
>> to allow for hand control over sound localization, much like you appear to
>> be using the wii in the video. This worked well, except for the fact that
>> the user had to be positioned in a fairly confined area. The mobile device
>> control removes this restriction.
>> 
>> Last year we used Unity and the Oculus with the LeapMotion controller to
>> create a virtual room and move ‘grabable’ sound sources around it. We tried
>> providing the audio over headphones using Unity’s capabilities, and also
>> Immergo (DBAP) over speakers. Interestingly the test subjects preferred the
>> external speakers.
>> 
>> Yes, I have used the phone’s orientation capabilities to indicate a
>> sound’s localization, but don’t have a video of it. You can see the layout
>> on the phone here:
>> 
>> https://www.immersivedsp.com/ <https://www.immersivedsp.com/>
>> 
>> Checkboxes at the bottom select the various axes. The purple circle can be
>> touch-moved for two dimensions, and the white rectangle indicates
>> height.The blue circles indicate already recorded tracks that move with the
>> playback of the sound.
>> 
>> Would love to make the London demo and if possible will contact you,
>> thanks!
>> 
>> 
>>> On 30 Jun 2017, at 10:39 AM, Augustine Leudar <augustineleu...@gmail.com>
>> wrote:
>>> 
>>> by the way have you already done this thing pointing the phone and
>>> indicating a sound in 3d space - if so do you have a link to it ?
>>> This is a pretty crappy video of very early prototype I did a few years
>> ago
>>> - but things have moved on a lot since this :
>>> 
>>> https://www.youtube.com/watch?v=7cmodvSM5jE
>>> 
>>> On 30 June 2017 at 09:33, Richard Foss <r.f...@ru.ac.za> wrote:
>>> 
>>>> That sounds great Augustine, and fun! Certainly highlights the
>> flexibility
>>>> and capability of Max/MSP.
>>>> 
>>>> Immergo uses the gyroscope of the smartphone/tablet for orientation
>>>> determination.
>>>> 
>>>> The typical mode of working is to touch-move a sound source (circle)
>>>> around a representation of the sound space on the screen, for two
>>>> dimensional movement, and to tilt the device at the same time for the
>> third
>>>> dimension. You can request that all three axes are provided by phone
>>>> orientation though. So then it would be like pointing the device to
>>>> indicate the location of the sound.
>>>> 
>>>> The rendering algorithm is selectable - DBAP is used for unusual speaker
>>>> arrangements, like an art installation ’tunnel’ that we are currently
>>>> setting up. But you can select VBAP if the speakers are arranged on a
>>>> sphere. Big thank you’s to Trond Lossius and Ville Pulkki who made their
>>>> algorithms freely available!!
>>>> 
>>>> A long overdue and pleasurable task is to use Max with Immergo.
>> Currently
>>>> using the Reaper DAW. I will keep you updated.
>>>> 
>>>> 
>>>>> On 29 Jun 2017, at 11:38 PM, Augustine Leudar <
>> augustineleu...@gmail.com>
>>>> wrote:
>>>>> 
>>>>> Thats pretty impressive Richard - I would definately consider this
>> system
>>>>> in future for indoor stuff. In terms of interactive and automation - I
>> do
>>>>> that in MaxMSP mostly - so we have things like an accelerometer on your
>>>>> hand and wherever you point t

Re: [Sursound] Multichannel players for permanent installations

2017-06-30 Thread Richard Foss

That’s an inspiring video, thanks!

And many points of contact. A few years back we used the Microsoft Kinect to 
allow for hand control over sound localization, much like you appear to be 
using the wii in the video. This worked well, except for the fact that the user 
had to be positioned in a fairly confined area. The mobile device control 
removes this restriction.

Last year we used Unity and the Oculus with the LeapMotion controller to create 
a virtual room and move ‘grabable’ sound sources around it. We tried providing 
the audio over headphones using Unity’s capabilities, and also Immergo (DBAP) 
over speakers. Interestingly the test subjects preferred the external speakers.

Yes, I have used the phone’s orientation capabilities to indicate a sound’s 
localization, but don’t have a video of it. You can see the layout on the phone 
here:

https://www.immersivedsp.com/ <https://www.immersivedsp.com/>

Checkboxes at the bottom select the various axes. The purple circle can be 
touch-moved for two dimensions, and the white rectangle indicates height.The 
blue circles indicate already recorded tracks that move with the playback of 
the sound.

Would love to make the London demo and if possible will contact you, thanks!


> On 30 Jun 2017, at 10:39 AM, Augustine Leudar <augustineleu...@gmail.com> 
> wrote:
> 
> by the way have you already done this thing pointing the phone and
> indicating a sound in 3d space - if so do you have a link to it ?
> This is a pretty crappy video of very early prototype I did a few years ago
> - but things have moved on a lot since this :
> 
> https://www.youtube.com/watch?v=7cmodvSM5jE
> 
> On 30 June 2017 at 09:33, Richard Foss <r.f...@ru.ac.za> wrote:
> 
>> That sounds great Augustine, and fun! Certainly highlights the flexibility
>> and capability of Max/MSP.
>> 
>> Immergo uses the gyroscope of the smartphone/tablet for orientation
>> determination.
>> 
>> The typical mode of working is to touch-move a sound source (circle)
>> around a representation of the sound space on the screen, for two
>> dimensional movement, and to tilt the device at the same time for the third
>> dimension. You can request that all three axes are provided by phone
>> orientation though. So then it would be like pointing the device to
>> indicate the location of the sound.
>> 
>> The rendering algorithm is selectable - DBAP is used for unusual speaker
>> arrangements, like an art installation ’tunnel’ that we are currently
>> setting up. But you can select VBAP if the speakers are arranged on a
>> sphere. Big thank you’s to Trond Lossius and Ville Pulkki who made their
>> algorithms freely available!!
>> 
>> A long overdue and pleasurable task is to use Max with Immergo. Currently
>> using the Reaper DAW. I will keep you updated.
>> 
>> 
>>> On 29 Jun 2017, at 11:38 PM, Augustine Leudar <augustineleu...@gmail.com>
>> wrote:
>>> 
>>> Thats pretty impressive Richard - I would definately consider this system
>>> in future for indoor stuff. In terms of interactive and automation - I do
>>> that in MaxMSP mostly - so we have things like an accelerometer on your
>>> hand and wherever you point there is a sound etc etc - you also have a
>> map
>>> of the speakers and pan around that map (using Dbap basically) - Ive
>> seen a
>>> lot of third party systems that do automation etc - but I am already
>> pretty
>>> happy with max for interactive 3D audio - theres nothing Ive found I
>>> couldnt do yet ! But I guess I could integrate max easily enough with the
>>> AVB system.
>>> 
>>> On 29 June 2017 at 22:28, Richard Foss <r.f...@ru.ac.za> wrote:
>>> 
>>>> One other thing - you mentioned wanting to move audio channels in real
>>>> time - you can move up to 32 channels around in the speaker space,
>> using a
>>>> client-server program called Immergo. The server runs on the Mac and the
>>>> client on the Mac or a remote tablet - its web browser based, so you
>> type
>>>> in a URL. You successively select tracks and move them around a
>>>> representation of your speaker space. You can record and play back the
>>>> movements. The movements are realized via mixer matrices in the
>> speakers.
>>>> 
>>>> So its two different approaches you can use - pre-prepared channels with
>>>> movement incorporated (guess you would use the terminology ‘bed
>> channels’)
>>>> or channels that can be moved via the mixer matrices (object audio
>> style),
>>>> where metadata containing time stamped 3D coor

Re: [Sursound] Multichannel players for permanent installations

2017-06-30 Thread Richard Foss

That sounds great Augustine, and fun! Certainly highlights the flexibility and 
capability of Max/MSP.

Immergo uses the gyroscope of the smartphone/tablet for orientation 
determination.

The typical mode of working is to touch-move a sound source (circle) around a 
representation of the sound space on the screen, for two dimensional movement, 
and to tilt the device at the same time for the third dimension. You can 
request that all three axes are provided by phone orientation though. So then 
it would be like pointing the device to indicate the location of the sound.

The rendering algorithm is selectable - DBAP is used for unusual speaker 
arrangements, like an art installation ’tunnel’ that we are currently setting 
up. But you can select VBAP if the speakers are arranged on a sphere. Big thank 
you’s to Trond Lossius and Ville Pulkki who made their algorithms freely 
available!!

A long overdue and pleasurable task is to use Max with Immergo. Currently using 
the Reaper DAW. I will keep you updated.


> On 29 Jun 2017, at 11:38 PM, Augustine Leudar <augustineleu...@gmail.com> 
> wrote:
> 
> Thats pretty impressive Richard - I would definately consider this system
> in future for indoor stuff. In terms of interactive and automation - I do
> that in MaxMSP mostly - so we have things like an accelerometer on your
> hand and wherever you point there is a sound etc etc - you also have a map
> of the speakers and pan around that map (using Dbap basically) - Ive seen a
> lot of third party systems that do automation etc - but I am already pretty
> happy with max for interactive 3D audio - theres nothing Ive found I
> couldnt do yet ! But I guess I could integrate max easily enough with the
> AVB system.
> 
> On 29 June 2017 at 22:28, Richard Foss <r.f...@ru.ac.za> wrote:
> 
>> One other thing - you mentioned wanting to move audio channels in real
>> time - you can move up to 32 channels around in the speaker space, using a
>> client-server program called Immergo. The server runs on the Mac and the
>> client on the Mac or a remote tablet - its web browser based, so you type
>> in a URL. You successively select tracks and move them around a
>> representation of your speaker space. You can record and play back the
>> movements. The movements are realized via mixer matrices in the speakers.
>> 
>> So its two different approaches you can use - pre-prepared channels with
>> movement incorporated (guess you would use the terminology ‘bed channels’)
>> or channels that can be moved via the mixer matrices (object audio style),
>> where metadata containing time stamped 3D coords is recorded.
>> 
>> Richard.
>> 
>>> On 29 Jun 2017, at 10:52 PM, Richard Foss <r.f...@ru.ac.za> wrote:
>>> 
>>> That’s correct Augustine - just the Ethernet out the Mac. That will go
>> into an Ethernet AVB compliant PoE+ switch (extreme, netgear etc).
>>> 
>>> The SPK-4Ps will be connected to other ports of the switch. You can link
>> together switches for larger port numbers.
>>> 
>>> 
>>> 
>>> 
>>>> On 29 Jun 2017, at 10:25 PM, Augustine Leudar <
>> augustineleu...@gmail.com> wrote:
>>>> 
>>>> Hi Richard -
>>>> So if I have this right no interface is required - just the ethernet
>> out of
>>>> the mac ?
>>>> 
>>>> On 29 June 2017 at 20:49, Richard Foss <r.f...@ru.ac.za > r.f...@ru.ac.za>> wrote:
>>>> 
>>>>> Hi Augustine, to answer this question about the SPK-4Ps:
>>>>> 
>>>>>> can you daisy chain them (as discrete channels) or would you have to
>> run
>>>>> a seperate ethernet cable to each one?
>>>>> 
>>>>> You would run a separate Ethernet cable from a switch to each one. So
>> for
>>>>> example, running a DAW on a Mac, you would output your tracks to the
>>>>> outputs made available by the Mac AVB virtual audio entity, and each
>>>>> speaker would listen to one of the outputs.
>>>>> 
>>>>> You would use PoE+, not PoE, and each speaker is powered by a Merus
>> audio
>>>>> Eximo amplifier, providing 30W. With the Mac you could connect up to
>> 120
>>>>> speakers. You also have the possibility of connecting a separate
>> passive
>>>>> speaker to the powered one, providing 2 x 15W.
>>>>> 
>>>>> Here is a bit more info:
>>>>> 
>>>>> http://www.technologyintegrator.net/article/immersivedsp-
>>>>> integrates-merus-audio-eximo-amp-minidsp-spk-4p-speaker/

Re: [Sursound] Multichannel players for permanent installations

2017-06-29 Thread Richard Foss

One other thing - you mentioned wanting to move audio channels in real time - 
you can move up to 32 channels around in the speaker space, using a 
client-server program called Immergo. The server runs on the Mac and the client 
on the Mac or a remote tablet - its web browser based, so you type in a URL. 
You successively select tracks and move them around a representation of your 
speaker space. You can record and play back the movements. The movements are 
realized via mixer matrices in the speakers.

So its two different approaches you can use - pre-prepared channels with 
movement incorporated (guess you would use the terminology ‘bed channels’) or 
channels that can be moved via the mixer matrices (object audio style), where 
metadata containing time stamped 3D coords is recorded.

Richard.

> On 29 Jun 2017, at 10:52 PM, Richard Foss <r.f...@ru.ac.za> wrote:
> 
> That’s correct Augustine - just the Ethernet out the Mac. That will go into 
> an Ethernet AVB compliant PoE+ switch (extreme, netgear etc).
> 
> The SPK-4Ps will be connected to other ports of the switch. You can link 
> together switches for larger port numbers.
> 
> 
> 
> 
>> On 29 Jun 2017, at 10:25 PM, Augustine Leudar <augustineleu...@gmail.com> 
>> wrote:
>> 
>> Hi Richard -
>> So if I have this right no interface is required - just the ethernet out of
>> the mac ?
>> 
>> On 29 June 2017 at 20:49, Richard Foss <r.f...@ru.ac.za 
>> <mailto:r.f...@ru.ac.za>> wrote:
>> 
>>> Hi Augustine, to answer this question about the SPK-4Ps:
>>> 
>>>> can you daisy chain them (as discrete channels) or would you have to run
>>> a seperate ethernet cable to each one?
>>> 
>>> You would run a separate Ethernet cable from a switch to each one. So for
>>> example, running a DAW on a Mac, you would output your tracks to the
>>> outputs made available by the Mac AVB virtual audio entity, and each
>>> speaker would listen to one of the outputs.
>>> 
>>> You would use PoE+, not PoE, and each speaker is powered by a Merus audio
>>> Eximo amplifier, providing 30W. With the Mac you could connect up to 120
>>> speakers. You also have the possibility of connecting a separate passive
>>> speaker to the powered one, providing 2 x 15W.
>>> 
>>> Here is a bit more info:
>>> 
>>> http://www.technologyintegrator.net/article/immersivedsp-
>>> integrates-merus-audio-eximo-amp-minidsp-spk-4p-speaker/ <http://www 
>>> <http://www/>.
>>> technologyintegrator.net/article/immersivedsp- 
>>> <http://technologyintegrator.net/article/immersivedsp->
>>> integrates-merus-audio-eximo-amp-minidsp-spk-4p-speaker/>
>>> 
>>> Richard.
>>> 
>>> 
>>> 
>>>> On 29 Jun 2017, at 6:38 PM, Augustine Leudar <augustineleu...@gmail.com>
>>> wrote:
>>>> 
>>>> aye - and that would kind of defeat the advantage of it - Ill stick to
>>>> passive for now - Ive always found active speakers a bit of a pain for
>>>> larger site specific stuff because you have to run power to  them as well
>>>> as audio.
>>>> These look like they might be OK for some stuff :
>>>> 
>>>> https://www.immersivedsp.com/products/spk-4p
>>>> 
>>>> but Im a bit unsure as to what exactly they can do - can you daisy chain
>>>> them (as discrete channels) or would you have to run a seperate ethernet
>>>> cable to each one  ?
>>>> 
>>>> On 29 June 2017 at 17:04, Dave Hunt <davehuntau...@btinternet.com>
>>> wrote:
>>>> 
>>>>> Hi,
>>>>> 
>>>>> I think that you'll find that an active speaker with an AVB input will
>>>>> require more current than can be supplied over an ethernet cable.
>>> Perhaps
>>>>> for very small speakers with a low power digital amplifier, but anything
>>>>> decent would require mains (or DC batteries) to the speaker.
>>>>> 
>>>>> The available current may power the DAC though.
>>>>> 
>>>>> Ciao,
>>>>> 
>>>>> Dave
>>>>> 
>>>>> From: Augustine Leudar <augustineleu...@gmail.com>
>>>>>> Date: 28 June 2017 22:06:52 BDT
>>>>>> To: Surround Sound discussion group <sursound@music.vt.edu>
>>>>>> Subject: Re: [Sursound] Multichannel players for permanent
>>> installations
>>>>>> 
>>>>>> 
>>&

Re: [Sursound] Multichannel players for permanent installations

2017-06-29 Thread Richard Foss

That’s correct Augustine - just the Ethernet out the Mac. That will go into an 
Ethernet AVB compliant PoE+ switch (extreme, netgear etc).

The SPK-4Ps will be connected to other ports of the switch. You can link 
together switches for larger port numbers.




> On 29 Jun 2017, at 10:25 PM, Augustine Leudar <augustineleu...@gmail.com> 
> wrote:
> 
> Hi Richard -
> So if I have this right no interface is required - just the ethernet out of
> the mac ?
> 
> On 29 June 2017 at 20:49, Richard Foss <r.f...@ru.ac.za 
> <mailto:r.f...@ru.ac.za>> wrote:
> 
>> Hi Augustine, to answer this question about the SPK-4Ps:
>> 
>>> can you daisy chain them (as discrete channels) or would you have to run
>> a seperate ethernet cable to each one?
>> 
>> You would run a separate Ethernet cable from a switch to each one. So for
>> example, running a DAW on a Mac, you would output your tracks to the
>> outputs made available by the Mac AVB virtual audio entity, and each
>> speaker would listen to one of the outputs.
>> 
>> You would use PoE+, not PoE, and each speaker is powered by a Merus audio
>> Eximo amplifier, providing 30W. With the Mac you could connect up to 120
>> speakers. You also have the possibility of connecting a separate passive
>> speaker to the powered one, providing 2 x 15W.
>> 
>> Here is a bit more info:
>> 
>> http://www.technologyintegrator.net/article/immersivedsp-
>> integrates-merus-audio-eximo-amp-minidsp-spk-4p-speaker/ <http://www 
>> <http://www/>.
>> technologyintegrator.net/article/immersivedsp- 
>> <http://technologyintegrator.net/article/immersivedsp->
>> integrates-merus-audio-eximo-amp-minidsp-spk-4p-speaker/>
>> 
>> Richard.
>> 
>> 
>> 
>>> On 29 Jun 2017, at 6:38 PM, Augustine Leudar <augustineleu...@gmail.com>
>> wrote:
>>> 
>>> aye - and that would kind of defeat the advantage of it - Ill stick to
>>> passive for now - Ive always found active speakers a bit of a pain for
>>> larger site specific stuff because you have to run power to  them as well
>>> as audio.
>>> These look like they might be OK for some stuff :
>>> 
>>> https://www.immersivedsp.com/products/spk-4p
>>> 
>>> but Im a bit unsure as to what exactly they can do - can you daisy chain
>>> them (as discrete channels) or would you have to run a seperate ethernet
>>> cable to each one  ?
>>> 
>>> On 29 June 2017 at 17:04, Dave Hunt <davehuntau...@btinternet.com>
>> wrote:
>>> 
>>>> Hi,
>>>> 
>>>> I think that you'll find that an active speaker with an AVB input will
>>>> require more current than can be supplied over an ethernet cable.
>> Perhaps
>>>> for very small speakers with a low power digital amplifier, but anything
>>>> decent would require mains (or DC batteries) to the speaker.
>>>> 
>>>> The available current may power the DAC though.
>>>> 
>>>> Ciao,
>>>> 
>>>> Dave
>>>> 
>>>> From: Augustine Leudar <augustineleu...@gmail.com>
>>>>> Date: 28 June 2017 22:06:52 BDT
>>>>> To: Surround Sound discussion group <sursound@music.vt.edu>
>>>>> Subject: Re: [Sursound] Multichannel players for permanent
>> installations
>>>>> 
>>>>> 
>>>>> Hi Richard -
>>>>> There is lots of things that pan all around the room but the
>> soundscape is
>>>>> designed in pre production for this installation and then its just a
>>>>> multichannel file (or seperate mono ones). Other installations I do are
>>>>> interactive and so sounds are processed and panned in realtime around
>> the
>>>>> space depending on the sensors used etc. Certainly the AVB thing might
>> be
>>>>> good for some less inhospitable environments- I especially like the way
>>>>> power can be taken to the the speakers via the ethernet cable.
>>>>> 
>>>> 
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>>>> 
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>>>> Sursound@music.vt.edu
>>>> https://mail.music.vt.e

Re: [Sursound] Multichannel players for permanent installations

2017-06-28 Thread Richard Foss

Yes, I agree with Guillaume - AVB is a good option. miniDSP sell an 8 
channel box AVB box, the N-DAC8, for $299.


With three of these you have 24 channels, and they all run off the same 
network clock. You would also need a small (4/5 port) switch.


A more expensive, but also easily configurable option is their SPK-4P 
Power over Ethernet option ($395). For 22 channels, you would then have 
22 speakers connected to a 24 port switch. Power, audio, and control 
then all travel over one cable, which makes for easy configuration. 
Again, all speakers are synced to the same network master.


Richard.


On 2017-06-28 07:39 PM, Guillaume Le Nost wrote:

Aside from dante, AVB is another option for multichannel audio over ethernet.
It is now available as standard (and for free) in any Mac running a recent os, 
and several options for avb enabled amplifiers are available: biamp, crown, 
l-acoustics to list a few.
If converters are required, the Motu AVB series can help, and is relatively 
inexpensive.

Guillaume


On 28 Jun 2017, at 18:00, Jörn Nettingsmeier <netti...@stackingdwarves.net> 
wrote:


On 06/27/2017 07:52 PM, Augustine Leudar wrote:
see my post before last for what went wrong. How much are the Joecos ?

https://www.thomann.de/gb/joeco_bbr64_dante_blackbox_recorder.htm

However, if you go the Dante route, you might as well use an old Minimac with a 
virtual soundcard (if 32 ch out is enough) and an appropriate Dante converter. 
Should be cheaper, if you have the Mac lying around, and no less robust if the 
hardware is otherwise ok.


--
Jörn Nettingsmeier
De Rijpgracht 8, 1055VR Amsterdam, Nederland
Tel. +49 177 7937487

Meister für Veranstaltungstechnik (Bühne/Studio), Tonmeister VDT
http://stackingdwarves.net
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--
Professor Richard Foss
Computer Science Department
Rhodes University
Grahamstown 6140
South Africa

Tel: +27 46 6038294
Cell: +27 83 288 9354
email: r.f...@ru.ac.za

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Re: [Sursound] Ambisonic Mic Comparison

2017-06-26 Thread Richard Lee

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

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

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

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

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

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

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

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

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

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

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

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

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

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


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Re: [Sursound] DIY 4d soundsystem

2017-05-17 Thread Richard Foss

Another reasonably priced possibility is to set up an Ethernet AVB network. For 
48 speakers one could use 6 x N-DAC8s ($299 each) from miniDSP together with 
two MOTU AVB switches. miniDSP have also recently come out with a PoE Ethernet 
AVB capable speaker. This allows for incremental expansion of the immersive 
sound system.


--
Professor Richard Foss
Computer Science Department
Rhodes University
Grahamstown 6140
South Africa

Tel: +27 46 603 8294
Cell: +27 83 288 9354
email: r.f...@ru.ac.za




> On 17 May 2017, at 1:41 PM, Augustine Leudar <augustineleu...@gmail.com> 
> wrote:
> 
> An interface suggestion might be a Motu 24ao with 3 Behringer ADA8200 adats
> - that will give you 48 outs for around £1000. Somebody mentioned some
> daisy-chainable firewire device recently which might wook out cheaper - but
> I forget the make/model.
> 
> On 17 May 2017 at 05:50, Matthew Palmer <palme...@mymail.vcu.edu> wrote:
> 
>> hi group,
>> 
>> i am looking for any advice or suggestions about low-cost materials i could
>> use to build the columns of this 4d sound system <
>> http://demonicsweaters.com/2015/03/the-amazing-4d-sound-
>> system-is-a-nicola-tesla-inspired-fully-immersive-audio-experience/ >
>> 
>> i have bought 48 logitech z506's (https://www.logitech.com/en-
>> us/product/surround-sound-speaker-z506)and am working to construct a way
>> to
>> mount the 5 speakers around the sub (any suggestions here are also
>> appreciated) & build 16 3-speaker high columns that can fit in the back of
>> a pick-up truck.
>> 
>> the design objective is minimal cost, safe, modular, optimized for ease-of
>> setup & travel.
>> *my recent proposal to a friend of a speaker stand to start & a custom PVC
>> structure to finish was deemed unsafe.
>> 
>> thanks for any help.
>> - matt
>> palme...@vcu.edu
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>> 
> 
> 
> 
> -- 
> Augustine Leudar
> Artistic Director Magik Door LTD
> Company Number : NI635217
> Registered 63 Ballycoan rd,
> Belfast BT88LL
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Re: [Sursound] The BBC & Quadrophony in 1973

2017-01-10 Thread Richard

Very interesting


> I think I am correct in saying the BBC (H) and NRDC (45j) decided to
> co-operate rather than have yet another two competing systems on
> the 4-channel scene which at the time already had CD-4, UD-4, SQ and
> QS in the arena.  BBC/NRDC each modified their encoding "towards"
> each other and called it HJ.

As I recall, the BBC were getting bad reviews of the stereo produced by matrix 
H and discussed this with MAG who proposed a compromise. UHJ was a variation 
within the 5 parameters which define an Ambisonic encoding, mainly a reduction 
in the centre front phase shift compared with 45J, based on many listening 
tests. It was known internally as 35JA’, so you can guess there was a whole 
family of small variations. BBC HJ was defined as a set of tolerance zones on 
the locus, mainly defined with reference to pairwise panning, I think it 
coincided with the UHJ locus at four points , so in that sense it was 
compatible; the exact parameters of both variations were defined by MAG. In 
subsequent tests the BBC made, primarily using SFMs, they also used our 
encoders, so I suspect drama productions such as Gilgamesh and Inferno 
Revisited are actually UHJ.

Geoffrey


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Re: [Sursound] The BBC & Quadrophony in 1973

2017-01-07 Thread Richard Lee

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

http://ambisonia.com/

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

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

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

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

"Alice's Adventures In Wonderland".

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

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Re: [Sursound] Blue Ripple Sound & SN3D?

2017-01-02 Thread Richard Furse

Happy New Year!


An update on this:

We *did* switch from FuMa to SN3D at the end of last year and have renamed
the TOA plugins so they are now the "O3A" plugins. The renaming is mainly to
help with migration from existing projects.

This turned into quite a major release in the end ("Version Two") and we're
just tying up the loose ends now. We took the opportunity to update all the
GUIs (no more wood panels!) and have added a whole new toolkit ("O3A View")
for work with film, particularly 360 video for Virtual Reality. There are
some (confusing) screenshots here:
http://www.blueripplesound.com/story/view-screenshots-2017 

Best wishes,

--Richard


> -Original Message-
> From: Sursound [mailto:sursound-boun...@music.vt.edu] On Behalf Of
> Richard Furse
> Sent: 10 August 2016 11:54
> To: 'Surround Sound discussion group' <sursound@music.vt.edu>
> Subject: [Sursound] Blue Ripple Sound & SN3D?
> 
> Hi there!
> 
> 
> 
> Blue Ripple Sound are wondering about changing their TOA plugins from
> FuMa
> to SN3D (ACN), see http://www.blueripplesound.com/story/consultation-
> sn3d.
> Does anyone have strong feelings on this?
> 
> 
> 
> Opinions, rants and raves appreciated, but I'd like to avoid kicking off
> another format debate on this list - perhaps any interested folk could
> respond using the email link on the page?
> 
> 
> 
> Many thanks,
> 
> 
> 
> --Richard
> 
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[Sursound] New home for m/c toolkit, etc

2016-11-08 Thread Richard


Hello all,

I have "retired" from my formal involvement with the University of Bath 
(because of no longer doing anything that could be recognised as 
"research"), and my account there has terminated. I have a new personal 
website, which is thus the new home for the multi-channel toolkit and 
the impulse responses from the Darkside 50 detector:


http://www.rwdobson.com/mctools.html
http://www.rwdobson.com/sspaces/sciencespaces.html

Please update any links and other references you may have for these.


Richard Dobson
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Re: [Sursound] Blue Ripple Sound & SN3D?

2016-08-11 Thread Richard Furse

I was about to say that this is the sort of thing you get right once and
then never need to touch, but actually I did see a plugin out in the wild
recently with the conversion wrong!

On SN3D, my main worries are (a) losing compatibility with existing
material/projects/kit and (b) it's SN3D, not N3D (which we HAD been
considering and is used in MPEG-H). BUT - those are definitely very
manageable issues and well worth it if there's a strong enough consensus
that modern tools are all heading that way! What I'm really hoping doesn't
happen is that we change convention and then some amazing new toy emerges
using something different...

Thanks,

--Richard


> -Original Message-
> From: Sursound [mailto:sursound-boun...@music.vt.edu] On Behalf Of
> Dave Malham
> Sent: 10 August 2016 22:47
> To: Surround Sound discussion group
> Subject: Re: [Sursound] Blue Ripple Sound & SN3D?
> 
> Awww, come on, a bit of mental exercise is good for the soul! :-) :-)
> 
> Well, that said, (and this maybe a shock to those that know me) I'm
> gradually moving towards the view that FuMa may have had its day. The only
> remaining reason really is compatibility with with earlier materiel and
> Soundfield mics and that can be dealt with conversion code, so long as
> everything is properly documented.
> 
> Dave
> 
> On 10 August 2016 at 22:27, Fons Adriaensen <f...@linuxaudio.org> wrote:
> 
> > On Wed, Aug 10, 2016 at 11:53:33AM +0100, Richard Furse wrote:
> >
> > > Blue Ripple Sound are wondering about changing their TOA plugins from
> > FuMa
> > > to SN3D (ACN), see http://www.blueripplesound.
> > com/story/consultation-sn3d.
> > > Does anyone have strong feelings on this?
> >
> > Just do it.
> >
> > During the past months I've been explaining Ambisonics to some
> > people who are going to work on HOA based applications. All of
> > them are something like 20 years younger than FOA, and when they
> > think about Ambisonics theory that means arbitrary order. To them,
> > using anything but N3D, SN3D and ACN seems like a form of masochism.
> > Maybe because I made them work out X-axis rotation with FuMa in and
> > out as an exercise. None of them got it right.
> >
> > 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)
> >
> > ___
> > 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.
> >
> 
> 
> 
> --
> 
> As of 1st October 2012, I have retired from the University.
> 
> These are my own views and may or may not be shared by the University
> 
> Dave Malham
> Honorary Fellow, Department of Music
> The University of York
> York YO10 5DD
> UK
> 
> 'Ambisonics - Component Imaging for Audio'
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Re: [Sursound] External phantom battery source for TetraMic PPAc

2016-07-24 Thread Richard Lee

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

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

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

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

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

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

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

2016-07-04 Thread Richard Lee

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

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

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

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

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

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

2016-06-13 Thread Richard Lee

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

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

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

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

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

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

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

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

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

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

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

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

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[Sursound] Coefficients for Octagonal Decoder with CF

2016-05-26 Thread Richard Graham

Hi list!

Can anyone point me to a set of coefficients for an octagonal decoder with CF 
similar to the coefficients on Blue Ripple Sound?

http://www.blueripplesound.com/decoding 


All the best,

Ricky
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Re: [Sursound] Facebook spatial workstation

2016-05-24 Thread Richard

Limited as there’s no Windows support yet



Apparently Facebook is now getting in on ambisonic audio!

https://facebook360.fb.com/spatial-workstation/

I’m just downloading now.

Any thoughts?

Steve





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Re: [Sursound] YouTube now supports Ambisonics (warning....part advertisement..)

2016-05-10 Thread Richard

Don't forget the programs 'copysfx' and 'chxformat' in the CDP 
MultiChannel Toolkit enable users to tinker with channel masks, 
including setting to zero (if that helps), or of course to write a plain 
old multichannel WAVE file.


Richard Dobson
http://people.bath.ac.uk/masrwd/mctools.html





On 10/05/2016 20:30, Michael Chapman wrote:




Here's the problem  when writing a video file, FFMPEG appears to use
the channel_mask from the audio file even if a different channel_mask is
specified on the command line.   So even when I specified
"-channel_layout
4.0", the resulting MOV file still had the channel layout set to "quad",
which was then rejected by YouTube.



1)
You can (quite easily) use a hex editor (standard on most Linuxes) to edit
the *.wav.

Or even script this.

Seem to remember doing this about ten years, yawn, ago ...  (Could do some
digging!)

2)
TheSoX people were really nice.

A "sox -force_channel_mask 4.0" could be requested
though
a request "sox -force_channel_mask abcd" that would allow any mask would
be more 'democratic'.

Michael

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

2016-04-01 Thread Richard Lee

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

2016-03-30 Thread Richard

The interesting case here is panned sources between the speakers; where 
'equal power" implies the sound follows an arc of a circle centred on 
the listener (constant distance = constant power etc). However, the 
intuitive/vernacular understanding of panning is a linear path between 
the speakers - which would require a sound in the centre to be louder, 
in the manner of a Doppler shift effect. Conversely the naive linear 
arithmetic crossfade, as is well known, would make the sound clearly 
recede at the centre (by 3dB) as if on an ellipse. What do folks here 
(those who use stereo panpots in mixing etc)  ~expect~ the path between 
the speakers to (be perceived to) be?



On 29/03/2016 19:51, Peter Lennox wrote:

wasn't the original conception for stereo = 90 degrees, but 'hole in the 
middle' effects led to standardising on the narrower figure?
Dr. Peter Lennox
Senior Fellow of the Higher Education Academy
Senior Lecturer in Perception
College of Arts
University of Derby



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Re: [Sursound] Soundfield MK IV Disassembly

2016-03-29 Thread Richard Lee

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

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

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

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

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

3   Removing the tetrahedron allows the cone to come off.

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

My apologies for this Heath Robinson design from my mispent youth.
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Re: [Sursound] Furse-Malham to ACN conversion

2016-03-24 Thread Richard Furse

N3D/ACN

http://mpeg.chiariglione.org/standards/mpeg-h/3d-audio/dis-mpeg-h-3d-audio

Best wishes,

--Richard


> -Original Message-
> From: Sursound [mailto:sursound-boun...@music.vt.edu] On Behalf Of
> Sönke Pelzer
> Sent: 24 March 2016 11:55
> To: Surround Sound discussion group
> Subject: Re: [Sursound] Furse-Malham to ACN conversion
> 
> Hi Aaron,
> 
> I looked up some recent MPEG-H papers, but couldn't find information about
> their HOA channel ordering and normalization scheme.
> Could you please point me to these?
> 
> Thank you,
> Sönke
> 
> 
> 2016-03-24 7:25 GMT+01:00 Aaron Heller <hel...@ai.sri.com>:
> 
> > Martin,
> >
> > Note that while AmbDec can accommodate FuMa normalization on input, it
> > still makes the connections to jack in ACN order so the inputs will often
> > appear in Jack in ACN order (and never in FuMa order).  I say "often"
> > because the Jack API does not have any notion of the 'order' of an
> > application's ports.  To make matters worse, some jack control clients,
> > like qjackctl, sort the ports by name, so they appear in the GUI in FuMa
> > order, but if you simply do a 'bulk connect' you'll find the individual
> > connections were made in ACN order.  You really have to check the port
> > names carefully when connecting.
> >
> > The ADT can generate decoders in Faust with any combination of the the
> > channel order and normalization conventions that I'm aware of (FuMa,
> ACN,
> > MPEG-H, ...).  Those can be completed to VST and used in Reaper.
> >
> > There are also 3rd-order FuMa and 5th-order panners for Ambix/ACN (the
> > latter thanks to Florian Grond).   They are in the faust directory,
> > called ambi_panner_fms.dsp and  ambi_panner_5_ACN.dsp.  They have a
> built
> > in pink noise source with is useful for testing.
> >
> > It can also generate adapter matrices to convert signals between any of the
> > order and normalization conventions. An example of an Ambix to FuMa
> > converter is in ambix2fuma.dsp .
> >
> > Also  feel free to send me the ADT setup you're using, if you'd like me
> > to check it.
> >
> > Aaron
> >
> > On Wed, Mar 23, 2016 at 2:30 PM, Dave Malham
> <dave.mal...@york.ac.uk>
> > wrote:
> >
> > > Hi Martin (and Eric!),
> > >  One very simple thing I would do, before doing anything else, with
> > any
> > > system that's playing, as we say, silly bu..ers, is just to play a well
> > > localisable sound out of each speaker (on its own) in turn and check that
> > > (a) it's coming out of the speaker it should (all connections are
> > correct)
> > > and that it sounds like it's coming from the direction you think it
> > should
> > > (acoustics not too disruptive). If you really want to be picky, stick a
> > > soundfield type mic at the nominal centre point and check correct B
> > format
> > > signals are produced for each speaker location at the same time. Only
> > then
> > > start worrying about decoders, plugin connections and the rest. I once
> > > worked out that in a simple 1st order system driving a cube of speakers,
> > > there are 16 million ways of it going wrong, without counting individual
> > > component failures in amps, etc. Of course, lots of these ways of going
> > > wrong are self cancelling (*both* ends of speaker cable can be
> connected
> > > wrongly, cancelling out the polarity inversion, for instance) which is a
> > > darn good job otherwise our job would be near impossible. So, checking
> > the
> > > simple things first is a good way to avoid delving around the complex..
> > >
> > > Good luck
> > >  Dave
> > >
> > >
> > > On 23 March 2016 at 18:13, Eric Benjamin <eb...@pacbell.net> wrote:
> > >
> > > > "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
> > &

[Sursound] Rapture3D Universal SDK

2016-03-24 Thread Richard Furse

[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] grambilib~ for Pd

2016-03-11 Thread Richard Graham

Hi Aaron,

Thank you for taking the time to look at the source; good catch! I’ll fix those 
issues immediately.

Ricky

> On Mar 11, 2016, at 12:00 PM, sursound-requ...@music.vt.edu wrote:
> 
> Hi Ricky,
> 
> I took a quick look at grambipan.c.  For FuMa U you've written
> 
> 189:   (*APout4++) = sample1 * (2 * cosf(sample2)) * cosf(sample3) *
> cosf(sample3); //U
> 
> whereas the correct expression (found on Richard Furse's webpage, for
> example) is
> 
> U:cos(2A) cos(E) cos(E)
> 
> Please note that cos(2x) does not equal 2cos(x).  I see similar problems in
> other definitions.
> 
> 
> You might want to take a look at my Python library,
> symbolic_spherical_harmonics.  It derives and writes out expressions for
> the spherical harmonics in a number of different languages.
> 
> $ python SymYlm.py --spherical --FuMa --four_pi --translation c 2 2
> pow(cos(phi), 2)*cos(2*theta)
> 
> $ python SymYlm.py --cartesian --FuMa --four_pi --translation c 2 2
> pow(x, 2) - pow(y, 2)
> 
> Here's the git repo:
>   https://bitbucket.org/ajheller/symbolic_spherical_harmonics 
> <https://bitbucket.org/ajheller/symbolic_spherical_harmonics>
> 
> --
> Aaron (hel...@ai.sri.com <mailto:hel...@ai.sri.com>)
> Menlo Park, CA  US

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[Sursound] grambilib~ for Pd

2016-03-09 Thread Richard Graham

Hi all,

I’ve been working on a simple ambisonics library for Pd. 

Here’s v1:

https://github.com/rickygraham/grambilib 


I would love to hear your thoughts!

Ricky
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Re: [Sursound] Ambisonic Decoder Design Resources

2016-02-23 Thread Richard Graham

Folks, many thanks for all these incredibly helpful resources.
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Re: [Sursound] Ambisonic Decoder Design Resources

2016-02-20 Thread Richard Graham

Hi Archontis,

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

I have already developed some basic ambisonic externals for Pd 
(http://rickygraham.net/?p=176401730 <http://rickygraham.net/?p=176401730>) 
based on encoding and decoding technical notes from Blue Ripple Sound 
(http://www.blueripplesound.com/decoding 
<http://www.blueripplesound.com/decoding>), which provides coefficients for 
basic array set-ups. 

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

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

Many thanks,

Ricky

> On Feb 20, 2016, at 12:00 PM, sursound-requ...@music.vt.edu wrote:
> 
> Hi Richard,
> 
> there are numerous articles, do you intend to use anything more specific? 2D 
> or 3D decoding? regular or irregular setups? Or you are looking for the most 
> general case?
> (when you say in C, do you mean published code examples?)
> And by low-orders do you mean first-order systems mainly (b-format)? Most 
> practical systems at the moment target up to 3rd-order, which are not very 
> high orders either, but it makes a difference.
> 
> Regards,
> Archontis

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[Sursound] Ambisonic Decoder Design Resources

2016-02-20 Thread Richard Graham

Hi all,

Can anyone recommend literature on designing ambisonic decoders (in C)? 
Specifically, articles that speak to the calculation of coefficients for 
low-order decoders. 

All the best,

Ricky
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[Sursound] Servicing a Soundfield ST250

2016-02-15 Thread Richard Furse

Sorry if this is a bit off-topic...

Does anyone know where I can get a Soundfield ST250 serviced? It's working, but 
I'd like to make sure its calibration is optimal (I've a vague recollection 
there are some gains to tweak inside?). I've tried Soundfield already. I'm 
based in London.

Thanks,

--Richard


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

2016-01-29 Thread Richard Furse

Interesting thread!

At the risk of accusations of marketing (the shame, the shame!)*, I thought I'd 
provide a few details on how Blue Ripple Sound approaches binaural HOA 
decoders. We DO use an approach which shares quite a bit with Archontis' 
suggestions. I prefer this to the "virtual speaker" approach, but (as with many 
things HOA) there are lots of ways to get it badly wrong, and some of the 
maths/processing involved in the approach we settled on is some of the scariest 
I've done. The original Rapture3D binaural decoders (Orange, Yellow, Red, Blue, 
Green) were generated with our third generation of this tech (IIRC - something 
like that). Amber came out of our fourth generation (a rewrite) quite a few 
years later.

As of Rapture3D Advanced v2.7.0, we've been supporting import of AES69 personal 
HRTF files. This is our fifth generation - hopefully we now have this working 
sufficiently robustly to let it run automatically. That said, we've not 
publicised it widely - I'm not even sure I've posted anything here - that is 
because this is very definitely an "experimental" feature right now. AES69 is 
based on SOFA, which is a flexible (i.e. complex) file format - you need to be 
using HDF5 and NetCDF4 (not earlier versions) to get to the real data, and then 
there are a number of optional geometric transforms - we currently can't be 
100% sure we've handled all cases correctly. Then there's our scary processing, 
which obviously has a sensitivity to data quality - and we don't know what will 
be thrown at this code in the future. And finally, and probably most 
importantly, the current pool of testers is rather small because personal HRTF 
data is rare - we'd LOVE more feedback in this area - one day we'd lik
 e to remove the "experimental" tag. So, if you're measuring HRTFs, or 
estimating HRTFs (and I think that's the future) and are interested in HOA 
and/or Rapture3D Advanced, we'd love to hear from you.

On orders: doing this "perfectly" requires very high orders indeed, but if you 
do things right then results just get better as the order increases. At first 
order, the effect is there, and reasonably natural - but not strong (and 
head-tracking is easy enough, as Archontis mentions). Third order is nice and 
solid. Rapture3D Advanced internally spits out decoders up to seventh order, 
although the VST only supports first to third order. (We actually have an 
in-house version of all our VST plugins at seventh order but currently have no 
plans to release it - third order is PLENTY for content right now IMHO.) You 
can render at up to fifth order using our old Rapture3D Advanced OpenAL game 
engine. The new Rapture3D VR/game engine will not support AES69 import 
initially, for simplicity during deployment - although temporarily the beta is 
including my personal HRTF, which was imported from AES69, and we've included 
some hooks for the future...

Best wishes,

--Richard

[*] More seriously - please let me know off-list if I've misjudged this - I'm 
never sure about this sort of thing...


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

2016-01-29 Thread Richard Lee

Just to bring everyone down to earth ..

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

Greene/Lee Neckbrace
and
Malham/Van Gogh Experiment

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

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

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

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

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

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


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

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


CONCLUSIONS

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

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

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

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

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

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


SPECIAL OFFER

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

2016-01-21 Thread Richard Graham

Dave wrote: 

>  -KPi/2 to +KPi/2 for elevation since we generally
> regard horizontal as 0 degrees (or radians) with downward angles negative
> and upward angles as positive. 

Hi Dave,

Thanks, this does make sense. For anyone else interested:

Elevation for ambipan~ is defined as: x->APdELeft = (x->APfElevationLeft - 0.5) 
* kPI;
kPI = 3.14159265358979323846
x->APfElevationLeft = elevation input for, in this case, the left-most audio 
input
All input parameters are scaled 0 to 1. 
0 to 1 input offset by -0.5 gives us -0.5 to 0.5 range. 
Multiply that by 3.141 give us -KPi/2 to +KPi/2 (-1.5705 to + 1.5705)

Great! Thanks.

All the best,

Ricky

> 0 to 1 - 0.5)  * kPI.
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[Sursound] ambilib~ in C

2016-01-18 Thread Richard Graham

Hi all,

I have some quick questions; I’m writing my own audio-rate control version of 
the ambilib~ library for Pd / Max and I wanted to clarify the following:

The elevation param has an offset of -0.5: (x->APfElevationRight - 0.5)  * kPI. 

1) Why is that? 

2) The unit for elevation is radians, yes?

All the best,

Ricky


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Re: [Sursound] The Mike Skeet Collection

2015-12-22 Thread Richard G Elen


Hello all...

Regarding Mike Skeet's collection...

Courtesy of Tony Flynn, who says: "here is a complete list of Mike's 
things, excluding the home made kit, of which there is loads, and I 
don't know what's there."


If you are interested in any of these items, or you have any proposals 
on the archiving of Mike Skeet's recordings, please contact Tony 
directly: sunsetbridge [at] outlook.com


AUDIO TECHNICA
BP407 1L Long Shotgun £250.
BEYER
M130N Figure Of Eight Double Ribbon X1 £200.
BRUEL (DPA) 4006 TL X3 £700 each.
PEARL DS60 £1,850
SCHOEPS
CCM2 Omnis X4 £600 each.
CCM8 Figure Of Eight X3 £600 each.
CCM4V Side Address Cardioids X2 £600 each.
CCM4 Cardioid X3 £600 each.
CCM22 Open Cardioid X2 £600 each.
SENNHEISER
MKH40 Cardioid X6 £600 each.
MKH30 Figure Of Eight X5 £600 each.
MKH800 Twin Dual Cardioid Capsules X1 £1,500

NEUTRIK MINI SPL X1 £100
Motu 8X8 MK1 Audio Interface £70
Edirol R44 recorder £200
Edirol R88 recorder £700
Edirol R4 Recorder £700
TC Finalizer £250
TC M2000 £200

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[Sursound] The Mike Skeet Collection

2015-12-20 Thread Richard G Elen


Hi, everyone...

I heard yesterday from my friend John Whiting that the veteran recording 
engineer specialising in stereo and spatial sound, Mike Skeet, has 
passed away. I knew him from the days of my Editorship of Studio Sound, 
where he made several contributions, and over many years he recorded and 
privately released an extensive collection of unique recordings 
including binaural, Ambisonic and other surround-sound material that 
makes an important contribution to our understanding of spatial sound 
recording.


Mike leaves an extensive collection of master tapes and equipment, much 
of the latter specially built, that has nowhere to go. John Whiting put 
it thus:


...the master tapes of all the dozens if not hundreds of fine compact 
cassettes and then CDs that he produced over the years will probably 
end up on a scrap heap. A close professional friend has been trying to 
find a home for them and no one is interested, all the way up to the 
National Sound Archive. None of his surviving familiar is likely to 
give it house room. Even most of his sound equipment, much of it 
unique and all of it first rate, has over the last few months proved 
to be unsaleable and even unwanted as a gift.


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


I will happily pass on any suggestions to John Whiting - I am not sure 
whether he is on this list.


Many thanks,
--Richard Elen
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Re: [Sursound] The Mike Skeet Collection

2015-12-20 Thread Richard G Elen


Hi, John...

Yes, I am in touch with Heather on this topic.

Thanks for the offer, which I've passed on and I expect someone will be 
in touch.


-_R

On 12/20/2015 4:31 PM, John Leonard wrote:

Is it worth contacting the AES about his archive? Heather Lane knew Mike quite 
well until they lost touch and she’s aware of his passing and may be able to 
offer suggestions.

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Re: [Sursound] WFS in a small box?

2015-11-29 Thread Richard Lee

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

2015-11-23 Thread Richard

No, Ambisonics existed well before they came onto the scene, the word 
describing the process was used on the LP released which contained an Ambisonic 
recording (actually 45j) from 1974.

I don't believe the N.R.D.C was in existance then.

Richard


  I’ve just looked at the Wikipedia entry for Ambisonics, which contains this 
statement:

  "Ambisonics was developed in the UK in the 1970s under the auspices of the 
British National Research Development Corporation.”

  Well, I know they screwed up my little project because of their involvement, 
but is this really a correct statement of fact?

  Regards,

  John




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


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

2015-11-21 Thread Richard Dobson

I totally agree; and found much the same on many of the recent Bond 
films. 'Tomorrow Never Dies' was especially overladen and noisy; perhaps 
they felt they had to cover up the more than usually silly plot.
But I do have to acknowledge that the sound system in our local Frome 
cinema is probably not the best available. Film spatialisation seems far 
too often to involve routing this sound to that speaker. And it seems 
almost mandatory that all dialogue is on the Centre speaker. Yet there 
are commercial Ambisonic sound libraries, so where are the state of the 
art films which use them?


FWIW, there is a "sound design" group on Yahoo Groups (does come through 
as simple email), which is focussed on film sound. It's not a busy 
group, a trickle of messages from time to time. Currently there is a 
thread on foley, and how to record very quiet things. A few posts 
announcing sound libraries n'stuff. I have no idea if anyone is on there 
who is a film sound "A-list" person.


Richard Dobson


On 21/11/2015 08:09, Michael Chapman wrote:
...


Think I may have already told this one: Quite some years ago I had
insisted that the children watch videos in the original language _and_
without subtitles.
I caught them with the subtitles on.
They responded that the subtitles were in the same language as the
soundtrack, and they had them on because they couldn't hear the dialogue.
I listened for a bit ... and apologised.

Presumably there are special plug-ins that destroy sondtracks. It would be
fascinating if someone would (anonymously, I presume) come 'out' and share
the dirt ... ...




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

2015-11-21 Thread Richard Dobson

And yet another reason - with dialogue in glorious mono on one channel, 
dubbing for other languages is super-easy.


Richard Dobson

On 21/11/2015 10:52, Eero Aro wrote:
..

Ever since, 99% of all dialogue has been placed in the center channel
for the
reasons Dave is describing.

There are even more reasons. The timbre of the voice is different, if it
moves away
from the center channel.



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Re: [Sursound] Binaural broadcast

2015-10-28 Thread Richard

Many thanks for the heads up. I think i'd best warn folks hear in the K this 
will be best heard on the FM service, as the broadcasts via DAB, Freeview and 
Satellite use Jstereo to help reduce bandwidth usage. Apart from the damage it 
does to standard stereo material, it will seriously damage the binaural 
information.




  I believe one, possibly two, items due to be broadcast on BBC Radio 4
  on Saturday 31 October as part of a 'Fright Night' strand, were recorded
  in binaural stereo.

  22:00 GMT  "The Stone Tape"  A story by Nigel Kneale (he of 'Quatermass' 
  fame)

  23:00 GMT  "Ring" An adaptation of Koji Suzuki's 1991 novel.

  If the BBC website won't co-operate for non-UK residents,
  no doubt Expat Shield will spoof a UK IP for you.

  -- 
  Peter Carbines

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Re: [Sursound] SQ QUAD

2015-10-21 Thread Richard

Thank you for proving my point 

  --On 21 October 2015 18:16 +0200 Jörn Nettingsmeier
   wrote:

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

  Or maybe it won't fit in the margin of this email...

  Paul

  -- 
  Paul Hodges

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Re: [Sursound] SQ QUAD

2015-10-21 Thread Richard

I've done it, along with the QS one. Pretty awful really, he was spot on when 
he said that SQ was compatible with Ambisonics.

Of course i'd imagine that when he said it was better than SQ hardware decoders 
he was comparing it with a basic 10/40 non-logic decoder and certainly not 
against a Tate.

  David Pickett 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.

  MAG's distaste for SQ is well documented,
  and in December 1977 there was even a fight
  between him and Benjamin Bauer on the
  pages of New Scientist.  (This was refereed
  by Barry Fox (writing as Adrian Hope).)

  However, in July and August 1977 (refs at end),
  Gerzon published the design of an Ambisonic
  decoder that included an SQ mode (along with
  nine other modes).   In Part 1, Gerzon wrote:
 "SQ decoders cannot be designed to give
  full ambisonic results; there is even a
  mathematical theorem to this effect.  The
  decoder for SQ provided is, however, less
  phasey in quality than the SQ designs on
  the market, and was designed specifically
  for incorporation into this design.  It is not
  in accordance with CBS Laboratories' SQ
  specification, but in the author's opinion, it
  is better than decoders that are."

  In Part 2, the equations for decoding SQ are
  given as:
  W'' = 0.73*Sum
  X'' = -0.73*j*Sum
  Y'' = 0.73*Diff - 0.73*j*Diff
  where Sum = Left + Right
  and Diff = Left - Right

  As far as I can tell, W'' is the W' signal after
  the shelf filters (and the SQ mode did not use
  shelf filters).  Also the W' signal appears to be
  the W signal after removal of the Sqrt(2)
  weighting.  Anyone interested in implementing
  this decoder will need to read the refs, which
  will be somewhere in the Ambisonic
  Motherlode.

  Regards,
  Martin

  Michael Gerzon, "Multi-System Ambisonic
  Decoder":
  Part 1: "Basic Design Philosophy",
  Wireless World, vol. 83 no. 1499,
  pp. 43-47 (1977 July).
  Part 2: "Main Decoder Circuits",
  Wireless World, vol. 83 no. 1500,
  pp. 69-73 (1977 Aug.).
  later parts never written & published.

  -- 
  Martin J Leese
  E-mail: martin.leese  stanfordalumni.org
  Web: http://members.tripod.com/martin_leese/
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Re: [Sursound] SQ QUAD

2015-10-21 Thread Richard

That is very true, and there never will be.

Spent to long on it to get it where they are now, and having had nothing but 
negativity from certain areas regarding my work i fail to see why i should 
share it with them.

I did once share my work, in fact you and i have talked in the past, and if you 
dig REALLY deep you'l find my earlly versions for SQ, which are at least 1000% 
better than those two programs) but as others knew better i've let them go 
their own way (which apears to be nowhere) and i've gone mine.

The proof is in the pudding, as they say.




  - don't feel that I have fallen into any trap, as I have never even tried 
to
  understand the quad matrix systems. I used surround sound in production
  in my work in the 1990's and none of the matrix systems did what I
  wanted.

  I look forward to read from your blog, when you really reveal what the
  "advanced and mysterious decoding systems" really keep inside.
  Right now, I couldn't find any practical solutions that anyone could use
  from your site. That is why people keep using the programmes that I
  linked to.

  Until then,

  cheers

  Eero
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Re: [Sursound] SQ QUAD

2015-10-21 Thread Richard

As i said, saying's one thing, actually doing is another.

There's a great deal more to decoding SQ & QS, my proces's are highly advanced
and gives results approaching that of the original four channel master, which
just just doing simple maths will not do (max 3db)

You have fallen into the same trap that everybody falls into, that of
misunderstanding these old matrix systems and the complexity that lies within
what appears to be a simple equation.

I was most fortunate to have a 'pen-pal' relationship with Ben while i was at
college in the early 1970's and as been said here a few posts ago, yes he did
bang on about SQ, but nobody actually bothered to really listen, and lost the
chance to learn how interesting/complex SQ really was.

Perhaps you should listen to a 'Pheonix' so you can hear just how good SQ
actualy was. The issue with the system (as well as QS) was that technology
wasn't up to the job at the time, which gace them the bad names they aquired.

But, we're in the 21st century now...

Richard wrote:
> The basis for their work appears to be the many inacurate sites
> filling the web with 'oh-so' wrong, ill informed inacurate
> information.

Well, this subject isn't much of my interests, but at least Stephan Hotto's
decoder claims to use exactly the equations you are citing:

Implemented Decoding Matrixes:

SQ (CBS):
LF = L
RF = R
LB = 0.707 * jL - 0.707 * R
RB = 0.707 * L - 0.707 * jR

http://www.hotto.de/software/quadrophonicmatrixdecoder.html

Apart from that, I don't think see anything very "complicated" about that.
(And I am very bad with mathematics.)

Multipliers mean different gains, the needed phase shifts are simply +/- 90
and +/- 180 degrees. There's plenty of phase-shifter plugins available
that do
the job.

Been there, done that in AudioMulch. Worked fine for me.

Now, if you'd like to go the gain riding (logic, as they were called)
path of the
analog decoders, that's where I raise my hands.

Eero
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Re: [Sursound] SQ QUAD

2015-10-21 Thread Richard

The basis for their work appears to be the many inacurate sites filling the web
with 'oh-so' wrong, ill informed inacurate information.

I've had a bash at altering 'Wikipedia' in some areas but it's a thankless task
trying to undo the masses of misinformation out there.

I've withdrawn into my little world watching as the world goes slowly
mad.

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] SQ QUAD

2015-10-20 Thread Richard

Alas it's far more complex than that, a quick look at the equation will tell 
you that:

Lt = Lf + (- j0.707Lb + 0.707Rb) 

Rt = Rf + (- 0.707Lb + j0.707 Rb )

  Many thanks,  Eero!  (I dont know why my search didnt turn these up...)

  I see that the transformation used in the first one is simple to 
  describe. 90 degree rotation is not simple, however!

  David

  At 20:37 20-10-15, Eero Aro wrote:
  >There's another:
  >
  >http://www-user.rhrk.uni-kl.de/~junglas/SQdecode/SQdecode.html
  >
  >Eero
  >
  >20.10.2015, 20:35, David Pickett kirjoitti:
  >>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] Modified SoundField Mk IV

2015-10-19 Thread Richard Lee

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

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

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

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

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

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

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

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

   Dave

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

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

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Re: [Sursound] Modified SoundField Mk IV

2015-10-18 Thread Richard Lee

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

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

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

... but I think his post should be titled "Where Beach Bums & Angels fear to 
Tread" :)
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Re: [Sursound] Google Files Trademark for ‘360-Degree Spherical Audio’ Software

2015-09-09 Thread Richard Dobson

It doesn't sound quite the most robust title, really - either 
tautologous or self-contradictory?



Richard Dobson

On 09/09/2015 18:01, mgra...@mstvp.com wrote:

From yesterday's tech news headlines: Google Files Trademark for
'360-Degree Spherical Audio' Software


http://www.omgchrome.com/google-dynamic-virtual-surround-sound-trademark

 Sadly, no technical details offered.


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Re: [Sursound] ambisonics audio for 360 film (VR)

2015-08-20 Thread Richard Furse

Hi there!

This is definitely a hot topic at the moment, and I thought folk might be
interested in what we (Blue Ripple Sound) have been up to in this area.
There's a news items at
http://www.blueripplesound.com/story/rapture3d-for-virtual-reality.

The executive summary of that is: there's a new version of the Rapture3D
game engine in the pipes that is simpler, faster, more portable, and
includes support for TOA playback (with re-pointing). For VR, this means
you can create complete 3D audio scenes with our TOA studio plugins which
can then be rendered to binaural, in a way that takes the user's head
position into account.

Best wishes,

--Richard


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Re: [Sursound] ambisonics audio for 360 film (VR)

2015-07-22 Thread richard

On 22/07/2015 11:38, Henk | Spook.fm wrote:

Dear Surround Sound Discussion Group,

...

My question is: Who can help us find the best libraries for
ambisonics audio to use in the 360 Video player.

So far we have come up with the following:

you can seem to play ambisonics files with these:
https://github.com/ComposersDesktop/CDP7/tree/b085aa141fa93c98ae137feae5e7c58d656ddfdd/dev/externals/mctools

and

http://www.mega-nerd.com/libsndfile/

Are these the same? which one would be best for this project?

As the author of MCTools I should probably offer an answer. They are
basically utilities for assembling, disassembling, recording and playing
multichannel files, including but not only AMB; and there they include
very vanilla encoding and decoding (regular layouts, no filtering, no
binaural). A simple circular (rotating, up to 2nd-order horizontal) pan
is available, but no tumble/rotate etc processes. Whereas libsndfile is
a comprehensive C library for reading and writing soundfiles in umpteen
formats. AMB support (in terms of recognising the file format; no
processing) was added some while ago - which is why AMB is now somewhat
supported in Adobe Audition. MCTools mostly uses my own minimalist
soundfile library portsf, limited to PCM WAVE and AIFF files (of
course including AMB).

The link above is for the sources, as included with the Open-Source CDP
distribution.

Binaries (OS X, Windows) for MCTools can be got here:

http://people.bath.ac.uk/masrwd/mctools.html

Richard Dobson

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Re: [Sursound] SoundField rental in Spain

2015-06-26 Thread Richard Lee

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

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

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

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

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

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

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

http://ambisonic.info/audio.html

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

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

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

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

2015-06-17 Thread Richard Lee

How is the Oktava Tetrahedral mike calibrated?

In what form are the calibration files?

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

http://www.oktavausa.com/ProductsPages/Ambient4DMic.html has no info at all.
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[Sursound] another tasty lookingmulti-channel dsp kit

2015-06-17 Thread Richard Dobson


Hello all,

AD have a new dual-core SHARC-based evaluation board (long URL):

http://www.analog.com/en/design-center/evaluation-hardware-and-software/evaluation-boards-kits/EVAL-ADSP-SC589.html

Offers two i/o modes, 4 in + 8 out, or 12 out; plus Ethernet, USB 2, 
etc. Some obvious applications come to mind...


Richard Dobson


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Re: [Sursound] another tasty lookingmulti-channel dsp kit

2015-06-17 Thread Richard Dobson


On 17/06/2015 10:12, Michael Chapman wrote:

Hello all,

AD have a new dual-core SHARC-based evaluation board (long URL):

http://www.analog.com/en/design-center/evaluation-hardware-and-software/evaluation-boards-kits/EVAL-ADSP-SC589.html



Looks interesting ...

The webpage won't give me a price (some browser problem ...), about how
much ?



There are two EZKIT versions, shown as $435 and $495:

http://www.analog.com/en/design-center/evaluation-hardware-and-software/evaluation-boards-kits/EVAL-ADSP-SC589.html#eb-buy

The latter has ICE stuff and extended software licence for something, 
probably not needed in most cases.


Richard Dobson




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

2015-05-19 Thread Richard Lee

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

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

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

Which Soundfield did you use?

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

What was the Decoder?

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

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

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

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

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

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

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


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

2015-05-17 Thread Richard Lee

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

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

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

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

It's so nearly there but not quite.

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

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

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

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

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

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

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

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

2015-05-14 Thread Richard Lee

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

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

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

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

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

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

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

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

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

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

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

2015-05-12 Thread Richard Lee

The strict answer to your question is in Gerzon's

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

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

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

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

Aaron Heller  Fons Adrieansen are your best bets.

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

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

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

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

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

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Re: [Sursound] Infra sound Sub bass

2015-04-24 Thread Richard Lee

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

This is the case ONLY with organs reproduced properly.

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

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

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

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

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

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

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

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

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

2015-04-23 Thread Richard Lee

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

It is a MUSICAL one.

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

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

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

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

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

2014-11-04 Thread Richard Furse

At the risk of marketing (I'm never sure how much is appropriate to send
here? Let me know opinions off list!), you might also want to consider the
commercial Blue Ripple Sound TOA plugins and Rapture3D decoder. The
Rapture3D decoder can handle irregular speaker layouts (though obviously
large gaps between speakers will limit what can be done in those
directions).

BTW, and not yet posted here: the latest release of the TOA Manipulators
adds the TOA Zoom plugin, which we think is very exciting. It allows an
entire third order soundfield to be moved around in space (this can also be
thought of as a listener perspective transform). There's a new third order
compressor too.

Best wishes,

--Richard

 -Original Message-
 From: Sursound [mailto:sursound-boun...@music.vt.edu] On Behalf Of Jörn
 Nettingsmeier
 Sent: 03 November 2014 16:21
 To: sursound@music.vt.edu
 Subject: Re: [Sursound] Ambisonics for children's Museum
 
 Hi Jun,
 
 [...]
 
 I would use a Linux box for decoding - most versatile, stable and cost
 effective. My decoder of choice is AmbDec, or you could try Matthias
 Kronlachner's AmbiX tools under Mac OS X.
 
 
 Regards,
 
 
 Jörn
 
 [...]


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Re: [Sursound] Ambisonics for children's Museum

2014-11-04 Thread Richard Furse

I'm quite embarrassed now! Of course, the Blue Ripple stuff mostly isn't
free...

 -Original Message-
 From: Sursound [mailto:sursound-boun...@music.vt.edu] On Behalf Of
 Matthias Kronlachner
 Sent: 04 November 2014 19:52
 To: sursound@music.vt.edu
 Subject: Re: [Sursound] Ambisonics for children's Museum

 On 03/11/14 17:21, Jörn Nettingsmeier wrote:
  I would use a Linux box for decoding - most versatile, stable and cost
  effective. My decoder of choice is AmbDec, or you could try Matthias
  Kronlachner's AmbiX tools under Mac OS X.
 The ambiX plug-ins work equally good under Windows :-)
 Quite some people are happy with them in combination with the Ambisonic
 Decoder Toolbox (https://bitbucket.org/ambidecodertoolbox/adt.git)

 Matthias

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Re: [Sursound] Splitting a 10.2 file

2014-09-05 Thread Richard Dobson

Unfortunately it not quite as easy as it should be, as companies such as 
Steinberg and Digidesign extend (i.e. abuse) the documented WAVE 
format with such things as over-large format chunks; I have had to 
modify my strictly compliant code more than once to accommodate such 
technically non-compliant files.


This why people writing soundfile code for critical production work 
really do need to use libsndfile, as the author has refined it over very 
many years (and of course it is used everywhere so tested to 
destruction, so to speak) to catch all such idiosyncracies. I have 
limited myself to WAVE and AIFF PCM (and of course AMB).


The channelx (CDP multi-channel toolkit, now open source so also 
available for Linux) command line is really very simple, as it is almost 
automatic, e.g. to split a 6-chan file hex.wav to files called 
hex_c*.wav:


channelx -ohexsplit.wav hex.wav 1 2 3 4 5 6

gives files called hexsplit_c1.wav ... hexsplit_c6.wav

(the -o flag option allows you to specify a custom base outfile name to 
which the '_cN' suffix will be added as needed).


see: http://people.bath.ac.uk/masrwd/mctools.html
Last updated March 2014


Richard Dobson





On 04/09/2014 01:48, Sampo Syreeni wrote:

On 2014-09-03, Martin Leese wrote:


It's a command line program. Very fast. Needs lotsa careful typing.


Or sloppy typing into a text editor to create a BAT file.  Then
execute the BAT file.


And, by the way, if you need some de novo code to do something like
that, it ain't gonna be too expensive in the first place.
IFF/AIFF/RIFF/QTFF/BMFF and their ilk are downright ridiculously easy to
decode and sieve. Let's say, it only takes a day's worth of effort to
extract what you want from them, so that you could easily offer $80/h
before taxes for the effort.

Encoding into the various forms, that's another ballgame.


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Re: [Sursound] [Bulk] Re: Spheric Collection: The Largest Ambisonic Surround Sound Library

2014-08-01 Thread Richard

Hmm, i'm sure we've been down this road before, but...

I guess they have a good way of converting B-Format to Binaural, anyone got a 
really good way of doing the reverse?

Richard

 
Le 2014-08-01 10:03, David Pickett a écrit :

Which are the B-Format samples?  All I can find are descrubed as
Binaural Virtual Surround Sound -- use headphones for best results.

All of them: the binaural playback is for Internet preview only.

- Daniel


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Re: [Sursound] BBC Proms in 4.0

2014-07-18 Thread Richard

Firstly, i must say a thank you for letting us know. But, after the last test
it is obvious that a large majority of us were unable to enjoy these broadcasts
due to the limited requirements.

May i ask a question as a BBC licence payer. Why is this not available via the
standard broadcast methods, Freeview Freesat?

The BBC will make the BBC Proms Concerts available in 4.0 using MPEG-DASH. The
stream will be available internationally.

This is an experiment and may not always be working, but please give feedback
through the blog or twitter using #BBCProms4

The blog with details and links to the player are here
http://www.bbc.co.uk/blogs/radio3/posts/BBC-Proms-in-Surround-Sound

_
Rupert Brun.
Head of Technology, BBC Radio.

5045 Broadcasting House
Portland Place
London W1A 1AA

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Re: [Sursound] And now for something different...

2014-06-23 Thread Richard Dobson

I think the AES already has a project to define a file format for htrfs; when I 
get home I can find the project code. 

Richard Dobson


Sent from my iPhone

On 23 Jun 2014, at 17:43, Martin Leese martin.le...@stanfordalumni.org wrote:

 Bo-Erik Sandholm wrote:
 
 Is there a way to get a personalized HRTF (or even one near mine) with out
 spending many hundreds of the coins of your choice or travelling to a
 distant destination?
 
 No but, if the Microsoft stuff works out, there
 might be.
 
 Is there a standard format for HRTFS that can be used in several softwares
 or even converted?
 
 The answer is, again, no.  However, to state
 the obvious, if HRTFs are going to fly then
 there needs to be.  Is this a task for the AES
 and/or the EBU?
 
 To continue stating the obvious, most
 audio-only listening currently takes place using
 ear-buds plugged into players or phones.  This
 doesn't look like it is going to change anytime
 soon.  Binaural with personalized HRTFs would
 improve this listening experience.
 
 Regards,
 Martin
 -- 
 Martin J Leese
 E-mail: martin.leese  stanfordalumni.org
 Web: http://members.tripod.com/martin_leese/
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