Re: [Sursound] Ambisonics decoder to hrtf with VR support

2015-03-13 Thread Alexis Shaw
> On 13 Mar 2015, at 10:19 pm, Jörn Nettingsmeier 
>  wrote:
> 
>> On 03/09/2015 12:12 PM, Tobix wrote:
>> 
>> I've read that ambisonics is good for listener in center, right? This
>> means that if player can move the sound effect will be distorted?
> 
> If you're using pre-rendered Ambisonics files, the listener will never move 
> from the sweet spot, translations are impossible. What you do is track the 
> rotations of the listener's head and rotate the rendering accordingly.
> 
> If you want to do translations, you will have to render the scene in realtime.
> It's very much like 3D cinema: you can produce fixed content for a 
> pre-defined viewpoint with a pair of spaced cams, but if you want to allow 
> the viewer to move, you need to model the whole scene.
> 
There are techniques that with HOA will give you some translation. That being 
what makes it higher order.
>> The way that openal handles source positions and listener is good for
>> me, but could it be reproduced with ambisonics?
> 
> Yes. Ambisonics can just as well be used as a realtime rendering format. But 
> there is a tradeoff: if the number of discrete sources is small compared to 
> the number of virtual speakers, direct rendering is cheaper.
> 
> Consider the case of a virtual 3rd-order 3D rig, let's assume an icosahedron. 
> The cost of decoding the 16ch B-format to 20 speaker feeds is negligible, but 
> you will have to convolve those with 20 pairs of HRTFs, tracked in realtime.
> 
You do realise that you don't have to use virtual speakers for the actual 
audio. If you take the impulse response of each Ambisonic channel and pass it 
through the chain, then you can convolve directly with that. (What with the 
linear, time invariance). That means that you have to do 20 FFTs, 
multiplication for filtering and 2 IFFTs. Not saying that this will end up 
faster in all cases but a good thing to note. 

> This rendering effort will be constant, regardless of the number of sound 
> sources in your scene. So if it's just a few, it's easier to just convolve 
> each source with the two HRTFs. At 20 sources, you're break-even, above that, 
> 3rd order Ambi is cheaper.
> 
> The situation changes a bit if you consider the diffuse field for 
> reverb/ambience: it can be mixed into the Ambi signal at no extra cost, but 
> if modeled with individual sources, it's expensive, because you need quite a 
> few.
> 
> 
> Best,
> 
> 
> Jörn
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Regards
Alexis. 
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Re: [Sursound] Dual hexagon using in-ceiling uppers and matching in-wall lowers

2013-12-16 Thread Alexis Shaw
On Tuesday, December 17, 2013, Fons Adriaensen wrote:

> On Tue, Dec 17, 2013 at 10:53:23AM +1100, Alexis Shaw wrote:
>
> > > 1. That's giant overkill, completely useless. Whatever you need
> > >as filtering in a decoder can be done easily with much shorter
> > >impulse response. Or even with some simple IIR filters.
> >
> > It is overkill for anything except room correction where you often need
> 1/2
> > second of filtering
> > At 386khz that leads to a 128k fir filter
>
> * 386 kHz ?? Even 192 is completely useless.
>
> * Half a second of room correction would be useful only at very low
>   frequencies (unless you want to correct your bathroom). And if and
>   when it is necesssary, that part of the frequency range can be
>   processed at a much lower sample rate and require only a short filter.
>
> * And then, room correction would be done on the speaker signals,
>   not on the complete matrix. The latter *is* possible of course,
>   and would be ideal, *iff* you can derive the filters. Which would
>   require measuring the room response in higher order Ambisonic format,
>   not on option unless you have a EigenMike, and even then dubious.
>
> My other project is making an eigenmike type microphone using 1000 mems
microphones
I am intending to do that

>
> > > 2. The method you propose (DFT, matrix, IDFT) is wrong, you'd need
> > >linear convolution which is not the same thing.
> > >
> > > No you are wrong here, convolusion in. The time domain is equivelent to
> > multiplication in the
> > Fourier domain. That is simple sampling theory. Look up the overlap and
> add
> > method of fir filter implementation
>
> You don't have to tell me that (as the author of zita-convolver).
> But DFT, multiply, IDFT without overlap (which is what the OP
> described) won't do it.


Btw I am the op.

>
> Ciao,
>
> --
> FA
>
> A world of exhaustive, reliable metadata would be an utopia.
> It's also a pipe-dream, founded on self-delusion, nerd hubris
> and hysterically inflated market opportunities. (Cory Doctorow)
>
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Re: [Sursound] Dual hexagon using in-ceiling uppers and matching in-wall lowers

2013-12-16 Thread Alexis Shaw
Hi Joseph.

I am thinking of hacking the emotiva UMC-200, if I use the secondary zone
DACs I should be able to get some reasonable quality out of it.

Regards
Alexis Shaw

On Tuesday, December 17, 2013, Joseph Anderson wrote:

> Hello Alexis,
>
> Sounds like a fun project. What is the hardware you're hacking?
>
>
> My best,
>
>
> 
> Joseph Anderson
>
> j.ander...@ambisonictoolkit.net 
> http://www.ambisonictoolkit.net
> ~~~~~~~~
>
>
>
> On 16 Dec 2013, at 3:13 pm, Alexis Shaw >
> wrote:
>
> > Hi Joseph
> >
> > In terms of decoder I was intending to hack a hifi preamp with an 8
> > input-12output matrix 128k long fir filters. That is DFT the inputs, for
> > each frequency in that fft multiply by a matrix, then do an IDFT on the
> > output.
> >
> > Of course you do something a little more complicated in order to fix up
> > latency problems.
> >
> > This, I think is the most general Ambisonic decoder that is possible at
> the
> > moment. And if you play with the coefficients you can also do room
> > correction, delay lines all in that framework.
> >
> > On Tuesday, December 17, 2013, Joseph Anderson wrote:
> >
> >> Hello Alexis,
> >>
> >> A few things
> >>
> >> A) How were you planning on running your decoder? Write code from the
> >> ground up? Build your own hardware?
> >>
> >> If you're looking for something somewhat between 'off the shelf' and
> 'grow
> >> your own', you may like to have a look at the Ambisonic Toolkit package
> for
> >> SuperCollider. From the web page, "SuperCollider is a programming
> language
> >> for real time audio synthesis and algorithmic composition."
> >>
> >> The new and the older pages are here:
> >>
> >> http://supercollider.github.io/
> >> http://supercollider.sourceforge.net/
> >>
> >>
> >> The page for the Ambisonic Toolkit is here:
> >>
> >> http://www.ambisonictoolkit.net
> >>
> >>
> >> B) On your decoder arrangement... I'd suggest you think about the
> >> bi-rectangle arrangement, which uses 8 loudspeakers. This can work well
> in
> >> a domestic situation. Four loudspeakers are placed in the horizontal
> plane
> >> (in a rectangle), and then two more on the ceiling and two on the floor.
> >> Imagine two planes bisecting each other: one horizontal and one
> vertical.
> >>
> >> The ATK has a wide variety of inbuilt decoders. This page lists them:
> >>
> >> http://doc.sccode.org/Classes/FoaDecoderMatrix.html
> >>
> >> For a bi-rectangle, you'd use the diametric decoder. For two hexagons,
> >> you'd use the periphonic decoder. The ATK also includes Near Field
> >> Compensation and Psychoacoustic Shelf Filtering, allowing you to
> implement
> >> classic, optimised decoders. Additionally, because SuperCollider is a
> >> programming language for audio synthesis and signal processing, you also
> >> get delay lines and multipliers (gain adjustment) to compensate for
> >> differences in loudspeaker distances.
> >>
> >>
> >> Hope this helps!!
> >>
> >>
> >> My best,
> >>
> >> 
> >> Joseph Anderson
> >>
> >> j.ander...@ambisonictoolkit.net  
> >> http://www.ambisonictoolkit.net
> >> 
> >>
> >>
> >>
> >> On 16 Dec 2013, at 1:25 am, Alexis Shaw 
> >> 
> >
> >> wrote:
> >>
> >>> Hello All
> >>>
> >>> I am a long time lurker here, and have been interested in setting up an
> >>> ambisonic listening room in my house for some time. I have finally got
> >> the
> >>> courage to try, however the only room that I have available to use is
> >>> multi-use and only of moderate size 20m^2 (3940*4750*2690mm) In order
> to
> >> be
> >>> able to make the most of the space I have been thinking about using
> >>> moderately high end in-wall and in ceiling speakers in an rectangular
> >>> double-hexagon arrangement.
> >>>
> >>> My question here is two fold:
> >>> 1 - Do you think that this is at all a good or interesting setup. Even
&g

Re: [Sursound] Dual hexagon using in-ceiling uppers and matching in-wall lowers

2013-12-16 Thread Alexis Shaw
On Tuesday, December 17, 2013, Fons Adriaensen wrote:

> On Tue, Dec 17, 2013 at 10:13:15AM +1100, Alexis Shaw wrote:
>
> > In terms of decoder I was intending to hack a hifi preamp with an 8
> > input-12output matrix 128k long fir filters. That is DFT the inputs, for
> > each frequency in that fft multiply by a matrix, then do an IDFT on the
> > output.
>
> 1. That's giant overkill, completely useless. Whatever you need
>as filtering in a decoder can be done easily with much shorter
>impulse response. Or even with some simple IIR filters.

It is overkill for anything except room correction where you often need 1/2
second of filtering
At 386khz that leads to a 128k fir filter

>
> 2. The method you propose (DFT, matrix, IDFT) is wrong, you'd need
>linear convolution which is not the same thing.
>
> No you are wrong here, convolusion in. The time domain is equivelent to
multiplication in the
Fourier domain. That is simple sampling theory. Look up the overlap and add
method of fir filter implementation

> 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
>
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Re: [Sursound] Dual hexagon using in-ceiling uppers and matching in-wall lowers

2013-12-16 Thread Alexis Shaw
Hi Joseph

In terms of decoder I was intending to hack a hifi preamp with an 8
input-12output matrix 128k long fir filters. That is DFT the inputs, for
each frequency in that fft multiply by a matrix, then do an IDFT on the
output.

Of course you do something a little more complicated in order to fix up
latency problems.

This, I think is the most general Ambisonic decoder that is possible at the
moment. And if you play with the coefficients you can also do room
correction, delay lines all in that framework.

On Tuesday, December 17, 2013, Joseph Anderson wrote:

> Hello Alexis,
>
> A few things
>
> A) How were you planning on running your decoder? Write code from the
> ground up? Build your own hardware?
>
> If you're looking for something somewhat between 'off the shelf' and 'grow
> your own', you may like to have a look at the Ambisonic Toolkit package for
> SuperCollider. From the web page, "SuperCollider is a programming language
> for real time audio synthesis and algorithmic composition."
>
> The new and the older pages are here:
>
> http://supercollider.github.io/
> http://supercollider.sourceforge.net/
>
>
> The page for the Ambisonic Toolkit is here:
>
> http://www.ambisonictoolkit.net
>
>
> B) On your decoder arrangement... I'd suggest you think about the
> bi-rectangle arrangement, which uses 8 loudspeakers. This can work well in
> a domestic situation. Four loudspeakers are placed in the horizontal plane
> (in a rectangle), and then two more on the ceiling and two on the floor.
> Imagine two planes bisecting each other: one horizontal and one vertical.
>
> The ATK has a wide variety of inbuilt decoders. This page lists them:
>
> http://doc.sccode.org/Classes/FoaDecoderMatrix.html
>
> For a bi-rectangle, you'd use the diametric decoder. For two hexagons,
> you'd use the periphonic decoder. The ATK also includes Near Field
> Compensation and Psychoacoustic Shelf Filtering, allowing you to implement
> classic, optimised decoders. Additionally, because SuperCollider is a
> programming language for audio synthesis and signal processing, you also
> get delay lines and multipliers (gain adjustment) to compensate for
> differences in loudspeaker distances.
>
>
> Hope this helps!!
>
>
> My best,
>
> ~~~~
> Joseph Anderson
>
> j.ander...@ambisonictoolkit.net 
> http://www.ambisonictoolkit.net
> 
>
>
>
> On 16 Dec 2013, at 1:25 am, Alexis Shaw >
> wrote:
>
> > Hello All
> >
> > I am a long time lurker here, and have been interested in setting up an
> > ambisonic listening room in my house for some time. I have finally got
> the
> > courage to try, however the only room that I have available to use is
> > multi-use and only of moderate size 20m^2 (3940*4750*2690mm) In order to
> be
> > able to make the most of the space I have been thinking about using
> > moderately high end in-wall and in ceiling speakers in an rectangular
> > double-hexagon arrangement.
> >
> > My question here is two fold:
> > 1 - Do you think that this is at all a good or interesting setup. Even
> with
> > lots of work. I have a moderate undergraduate level of understanding of
> DSP
> > and and have read almost everything that I can get my hands upon about
> both
> > spherical harmonics and ambisonics and so feel able to write a custom
> > decoder if I have to.
> >
> > 2 - If the answer to the first question is yes, do you think that the
> > Noble-Fidelity L-85 LRCS (in ceiling) and L-82 (in wall lowers) would be
> an
> > appropriate speaker for this purpose. If not what speaker would you
> > recomend (if any). I am not attached to these speakers however I cannot
> > afford much more than USD 300 for each speaker.
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> > https://mail.music.vt.edu/mailman/listinfo/sursound
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[Sursound] Dual hexagon using in-ceiling uppers and matching in-wall lowers

2013-12-16 Thread Alexis Shaw
Hello All

I am a long time lurker here, and have been interested in setting up an
ambisonic listening room in my house for some time. I have finally got the
courage to try, however the only room that I have available to use is
multi-use and only of moderate size 20m^2 (3940*4750*2690mm) In order to be
able to make the most of the space I have been thinking about using
moderately high end in-wall and in ceiling speakers in an rectangular
double-hexagon arrangement.

My question here is two fold:
1 - Do you think that this is at all a good or interesting setup. Even with
lots of work. I have a moderate undergraduate level of understanding of DSP
and and have read almost everything that I can get my hands upon about both
spherical harmonics and ambisonics and so feel able to write a custom
decoder if I have to.

2 - If the answer to the first question is yes, do you think that the
Noble-Fidelity L-85 LRCS (in ceiling) and L-82 (in wall lowers) would be an
appropriate speaker for this purpose. If not what speaker would you
recomend (if any). I am not attached to these speakers however I cannot
afford much more than USD 300 for each speaker.
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[Sursound] Spherical microphone array

2013-10-18 Thread Alexis Shaw
Hello all.

I was wondering what people thought of the usefulness of a 10cm
diameter spherical microphone array with 1000 mems detectors. (I'm Thinking
of using ADMP621s)

I know that this has a kr of about 3.2 but with the large number of
microphones I was thinking it might be possible to get a 4th order result
without significant spatial aliasing. Especially if The pattern was chosen
to minimise spatial aliasing.

In order to complement this I was thinking of increasing the frequency
response al lower frequencies with a second, open spherical array ala the
work of the university of sydney.

Do you think that this would be a worthwhile exercise.
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Re: [Sursound] Making a standalone 8ch player

2013-05-23 Thread Alexis Shaw
I'm working on having this implement a 4th order ambisonic decoder.

The cool thing here is that there is a dual core arm processor on board
that can
run linux, so I can have a 40 or more channel hardware interface and
have it act as
an output that takes a 4th order ambisonic signal from software. Or at
least that is
the idea.

There actually seems to be enough io to output to well over 100 channels.
And there
Is a heap of DSP resources on this thing.

I don't know how popular this would be or how much time it will take,
however there
Is a huge amount that can be done here.

On Thursday, May 23, 2013, Ben Bloomberg wrote:

> I have some FPGA code to drive 4 and 8 channel Burr Brown DACs (PCM3168a,
> PCM1608) I could pass along.
>
> It's quite messy and I haven't worked on it in a while (4 years) but it
> also implements a 3rd order ambisonic encoder/decoder and streaming input
> via USB. The coefficients are all stored in a LUT in onboard memory to
> avoid lots of multiplication/trig. It wouldn't be too hard to modify it to
> grab wavs from an SD card. I've been using the Nexys2 platform from
> digilent.  FPGALink is a pretty cool USB library that does highspeed IO.
> PCM3168 is a tough chip to solder though...
>
> Ben
>
>
>
> On Thu, May 23, 2013 at 8:47 AM, Alexis Shaw 
> >
> wrote:
>
> > I am working on a system that can drive 40 or so channels based on a zynq
> > 7020 processor.
> >
> > The zedboard which is the dev kit i am working with costs 320 for
> academics
> > and ~400 for commercial uses.
> >
> > This will then drive a series of dac boards that I am working on. They
> are
> > likely to cost about 200 each for 8 channels (ESS).
> >
> > Cheaper dac boards could be invisiged, or even direct driving digital
> class
> > D modulators. I am mainly working on the player-control hardware.
> >
> > Regards.
> >
> >
> > On Thursday, May 23, 2013, Augustine Leudar wrote:
> >
> > > Hi Alexis,
> > > yes the box would need DA converters if thats what you mean. I am
> > thinking
> > > 8 outputs to start with min 16 bit 44.1 but it would be nice to have
> > > something that could be easily customisable for more
> > >
> > > On 23 May 2013 14:08, Alexis Shaw 
> > > >
> > > wrote:
> > >
> > > >  Do you need internal dacs.
> > > >
> > > > How many channels do you need implemented.
> > > >
> > > > I am working on a solution to the same problem at the moment.
> > > >
> > > > On Thursday, May 23, 2013, Marc Lavallée wrote:
> > > >
> > > > > That's an interesting idea: clustering Raspberry Pis
> > > > > with cheap 8 channels usb modules and jackd2.
> > > > > --
> > > > > Marc
> > > > >
> > > > > "Michael Chapman"  
> > > > > >
> > a
> > > écrit :
> > > > >
> > > > > > > Hello all,
> > > > > > > I want to start making a standalone 8 channel player (maybe
> more)
> > >  -
> > > > > > > something that can be used in museums, festivals etc for sound
> > > > > > > installations that can just be turned on and will instantly
> start
> > > > > > > looping a
> > > > > > > multichannel composition on an sd card. At the moment I am
> using
> > > > > > > rather unwieldly setups of small computers and multi channel
> > > > > > > soundcards such as RME and motu. Cables can easily be jogged
> > loose
> > > > > > > and it would be nice to have something more robust and that
> staff
> > > > > > > can easily just turn on and off. So I have looked into the
> > arduino
> > > > > > > (only 12 bit audio) and the raspberry pi but neither seem
> > > > > > > suitable . Systems already avaailable are ludicrously expensive
> > > > > > > (1000s of euros) Has anyone got any ideas on the best way to go
> > > > > > > about this - is there something maybe Im missing with the
> > raspberry
> > > > > > > pi/ arduino that could be customised ? Perhaps a custom made
> > > > > > > circuit board ? Ideas ? best,
> > > > > > > Gus
> > > > > >
> > > > > > If you are thinking of wider applications I would really
> encourage
> > > > > > you to go modular ... that is daisy 

Re: [Sursound] Making a standalone 8ch player

2013-05-23 Thread Alexis Shaw
I am working on a system that can drive 40 or so channels based on a zynq
7020 processor.

The zedboard which is the dev kit i am working with costs 320 for academics
and ~400 for commercial uses.

This will then drive a series of dac boards that I am working on. They are
likely to cost about 200 each for 8 channels (ESS).

Cheaper dac boards could be invisiged, or even direct driving digital class
D modulators. I am mainly working on the player-control hardware.

Regards.


On Thursday, May 23, 2013, Augustine Leudar wrote:

> Hi Alexis,
> yes the box would need DA converters if thats what you mean. I am thinking
> 8 outputs to start with min 16 bit 44.1 but it would be nice to have
> something that could be easily customisable for more
>
> On 23 May 2013 14:08, Alexis Shaw >
> wrote:
>
> >  Do you need internal dacs.
> >
> > How many channels do you need implemented.
> >
> > I am working on a solution to the same problem at the moment.
> >
> > On Thursday, May 23, 2013, Marc Lavallée wrote:
> >
> > > That's an interesting idea: clustering Raspberry Pis
> > > with cheap 8 channels usb modules and jackd2.
> > > --
> > > Marc
> > >
> > > "Michael Chapman"  > a
> écrit :
> > >
> > > > > Hello all,
> > > > > I want to start making a standalone 8 channel player (maybe more)
>  -
> > > > > something that can be used in museums, festivals etc for sound
> > > > > installations that can just be turned on and will instantly start
> > > > > looping a
> > > > > multichannel composition on an sd card. At the moment I am using
> > > > > rather unwieldly setups of small computers and multi channel
> > > > > soundcards such as RME and motu. Cables can easily be jogged loose
> > > > > and it would be nice to have something more robust and that staff
> > > > > can easily just turn on and off. So I have looked into the arduino
> > > > > (only 12 bit audio) and the raspberry pi but neither seem
> > > > > suitable . Systems already avaailable are ludicrously expensive
> > > > > (1000s of euros) Has anyone got any ideas on the best way to go
> > > > > about this - is there something maybe Im missing with the raspberry
> > > > > pi/ arduino that could be customised ? Perhaps a custom made
> > > > > circuit board ? Ideas ? best,
> > > > > Gus
> > > >
> > > > If you are thinking of wider applications I would really encourage
> > > > you to go modular ... that is daisy chainable devices:
> > > >
> > > > 2+2+2+2 = 8
> > > > 4+4 = 8
> > > >
> > > > 2+2+2+2+ ... = "maybe more"
> > > >
> > > > even if it is
> > > > 8+ ... = "maybe more"
> > > >
> > > > Good hunting,
> > > >
> > > > Michael
> > > ___
> > > Sursound mailing list
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> > > https://mail.music.vt.edu/mailman/listinfo/sursound
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> > Sursound@music.vt.edu 
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>
>
>
> --
> 07580951119
>
> augustine.leudar.com
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Re: [Sursound] Making a standalone 8ch player

2013-05-23 Thread Alexis Shaw
 Do you need internal dacs.

How many channels do you need implemented.

I am working on a solution to the same problem at the moment.

On Thursday, May 23, 2013, Marc Lavallée wrote:

> That's an interesting idea: clustering Raspberry Pis
> with cheap 8 channels usb modules and jackd2.
> --
> Marc
>
> "Michael Chapman" > a écrit :
>
> > > Hello all,
> > > I want to start making a standalone 8 channel player (maybe more)  -
> > > something that can be used in museums, festivals etc for sound
> > > installations that can just be turned on and will instantly start
> > > looping a
> > > multichannel composition on an sd card. At the moment I am using
> > > rather unwieldly setups of small computers and multi channel
> > > soundcards such as RME and motu. Cables can easily be jogged loose
> > > and it would be nice to have something more robust and that staff
> > > can easily just turn on and off. So I have looked into the arduino
> > > (only 12 bit audio) and the raspberry pi but neither seem
> > > suitable . Systems already avaailable are ludicrously expensive
> > > (1000s of euros) Has anyone got any ideas on the best way to go
> > > about this - is there something maybe Im missing with the raspberry
> > > pi/ arduino that could be customised ? Perhaps a custom made
> > > circuit board ? Ideas ? best,
> > > Gus
> >
> > If you are thinking of wider applications I would really encourage
> > you to go modular ... that is daisy chainable devices:
> >
> > 2+2+2+2 = 8
> > 4+4 = 8
> >
> > 2+2+2+2+ ... = "maybe more"
> >
> > even if it is
> > 8+ ... = "maybe more"
> >
> > Good hunting,
> >
> > Michael
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Re: [Sursound] Patent application: Data structure for HOA

2012-11-01 Thread Alexis Shaw
For HRTF based sound, headphones work the best. The HRTF is the solution of
the in-head effects.

On 2 November 2012 14:07, Stefan Schreiber  wrote:

> Richard Dobson wrote:
>
>  The same is true of stereo too. There are people who just don't hear
>> stereo as stereo. If the response to "lack of perfection" is always "do
>> nothing", nothing will be done. Alternatively, if you use those generic
>> HRTFs, at least ~some~ people will be happy.
>>
>> BTW, the AES has just announced a project "AES-X212" to develop a file
>> format for HRTF data; "The format will be designed to include source
>> materials from different HRTF databases". See:
>>
>> http://www.aes.org/standards/**meetings/new-projects.cfm
>>
>>
>> Richard Dobson
>>
>
>
> The next and valid question is if stereo via headphones actually works so
> well at all... (Many people have problems, such as in-head effects, lack of
> perceived "real space", etc.)
>
> If you would fix these problems, then you could probably also reproduce
> convincing binaural surround via headphones.
>
> Best,
>
> Stefan Schreiber
>
>
>
>> On 31/10/2012 16:38, Martin Leese wrote:
>>
>>  Peter Lennox wrote:
>>>
>>>  Yes but...why not simply release stuff for mobiles in a generic
 binaural -
 skip the uhj altogether?

>>>
>>>
>>> Please, what is this "generic binaural"?
>>>
>>> Everyone has an individual HRTF.  If you
>>> release binaural recording using a generic
>>> HRTF then it will work for some and not for
>>> others.
>>>
>>> There have been attempts to systemise HRTFs,
>>> so that you set about four different parameters
>>> to produce an individual HRTF, but they never
>>> caught on.
>>>
>>> Regards,
>>> Martin
>>>
>>>
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Re: [Sursound] Patent application: Data structure for HOA

2012-10-30 Thread Alexis Shaw
Hi all

Listening here:
Except that all of us have different HRTFs. This means that while you can
convert UHJ to B-format and then to Binaural, the opposite is not true.

Secondly, with today's app enabled mobile phones a B-format player could be
released by anyone. Pop in a custom HRTF file and then you have full
periphonic sound on your mobile. Also being able to calibrate out the
phones.

Thirdly, there is no reason why ear-buds could not be good for this audio
especially in-ears. The big problem here is that there is a lot of DSP that
would have to be done on the fly. Given however how much is allready done
to decode the likes of AAC audio, it should not be intractable.

A B-format that can be used on mobile devices seems to be an interesting
idea. The problem seems to be infrastructure. Being able to tag an audio
stream as HOA and then automatically decode it seems to be the big problem.
I agree with the statement above that a big step would be to get a decoder
into VLC via ffmpeg. this would indeed seem to be a big first step. There
are published HOA decoder coefficients for 5.1 and it is relatively easy to
compute a Cube format array stream and then apply a HRTF to it. The work
just has to be done. Perhaps all that is really needed in terms of a format
is a way to tag an audio stream in Matroska, .ogg, .mp4 and avi as an
ambisonic stream instead of a normal multi-channel one, along with a fixed
channel order. Once in ffmpeg lots of other things come for free i.e.
firefox, chrome, VLC, mplayer, mplayer-classic, windows media player for
those with the ffmplayer codec installed. Someone just has to step up and
do it. The best option is not to define a compression technique, but to
define a audio layout and specify a channel order. If you pack 2 streams
into the file say with the UHJ stereo and stream 2 as the other two streams
with a different magic string as their designator, then you have a
backwards compatible playback format that can be decoded into WXYZ
ambisonic with little loss and recover the audio. The same is true if you
were to just render the audio as 2L = sqrt(2)W + Y and 2R = sqrt(2)W - Y
easy to get W and Y out, then just ship X and Z in another stream with a
custom tag.

Alexis Shaw

On 30 October 2012 21:24, Peter Lennox  wrote:

> Am I missing something? - for mobile use, wouldn't B-format to binaural be
> better than UHJ?
> Dr Peter Lennox
>
> School of Technology,
> Faculty of Arts, Design and Technology
> University of Derby, UK
> e: p.len...@derby.ac.uk
> t: 01332 593155
> 
> From: sursound-boun...@music.vt.edu [sursound-boun...@music.vt.edu] On
> Behalf Of Richard Lee [rica...@justnet.com.au]
> Sent: 30 October 2012 19:51
> To: 'Surround Sound discussion group'
> Subject: Re: [Sursound] Patent application: Data structure for HOA
>
> > Unless things have changed a lot, last I checked lossy compression
> messes up phase relationships, and that would be an issue for things like
> UHJ, which as long as portable stereo players with limited battery life
> (and thus limited CPUs), is the only viable, because stereo compatible,
> distribution format.
>
> > At this point in time, not only is most music listened on mobile
> devices, most music is even purchased on mobile devices, and that's
> strictly a stereo (or maybe binaural) world.
>
> Try this simple experiment.  Take your favourite Nimbus UHJ CD and rip it
> using the most evil MP3 encoder you can find .. probably the one built into
> the latest Windoz Media Player.
>
> Do this at 256kB/s and also (shock!  horror!) at 128kB/s.  Now listen to
> the resultant files on a mobile device.  Then you can pontificate to us on
> how the musicality has all escaped and no one is going to find these
> acceptable.
>
> You can also rip to a WAV file if your mobile device will play these and
> compare the MP3s with the 'original'.
>
> This is just testing Ronald's assertion about compressed UHJ on stereo
> mobile devices.  I dunno about full UHJ surround decode cos there don't
> seem to be any good ones in the public domain.
>
> PS  I expect you to hear ve.eery slight differenes with one MP3 and
> probably none with the other.  I won't insist on Double Blind bla bla but
> you might find that educational.
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