Re: [Sursound] Noise reduction on Ambisonic files

[Len Moskowitz]

David Worrall  wrote:

I have had much better luck with the TetraMic - as long as radio 
interference can be kept under control.


If you're having incidents of RFI with TetraMic, we recommend our latest 
phantom power adapter (PPA) system: PPAc.


With PPAc the unbalanced cable length can be kept to just a few inches, and 
RFI should be unusual except in extreme circumstances. The original PPA and 
PPA2 had longer lengths of unbalanced cable; they were RFI-resistant except 
in high-EMI areas.


Please feel free to contact me if you want to discuss this.


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


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Re: [Sursound] Noise reduction on Ambisonic files

[Garth Paine]

HI Paul

Have you tried this on an Ambisonic file?  I would be interested to hear your 
thoughts.

Cheers, Garth

On Aug 7, 2014, at 5:43 AM, Paul Hodges  wrote:

> --On 07 August 2014 13:41 +0300 Eero Aro  wrote:
> 
>> Many Sursounders may not be aware that there are practically
>> at all multichannel noise reduction systems available. They are all
>> stereo, and can process mono.
> 
> Adobe Audition can do noise reduction on multi-channel files.
> 
> Paul
> 
> -- 
> Paul Hodges
> 
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Re: [Sursound] Noise reduction on Ambisonic files

[Garth Paine]

Dear Eero - thanks for this piece of advice

On Aug 7, 2014, at 3:41 AM, Eero Aro  wrote:

> Just one more note Garth; When you have denoised the files,
> check that their length bitwise is the same as it used to be. At least
> in the large scale you are on the safe side then. Of course the
> phase may deviate during the file run, but I don't think that won't
> happen.

On another note, there are some very expensive 5.1 noise reduction tools, but 
they are way out of my reach 

Cheers,
Garth Paine
ga...@activatedspace.com

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Re: [Sursound] Noise reduction on Ambisonic files

[Garth Paine]

HI David

Yes all balanced into the Sound Devices 788T.  Cable length is short, 5 Meters 
and connects directly to the microphone with 4 XLR on the end 

Cheers, Garth


On Aug 7, 2014, at 12:52 AM, David Worrall  wrote:

> I never achieved satisfactory (electronically quiet) environmental recordings 
> with the portable Soundfield. Maybe the ones I used weren't properly 
> calibrated: they were certainly affected by humidity.
> I have had much better luck with the TetraMic - as long as radio interference 
> can be kept under control.
> Garth: what cable lengths are you using and are the inputs balanced?
> 
> D.
> 
> On 07/08/2014, at 1:03 AM, Eric Benjamin wrote:
> 
>> Garth,
>> 
>> I wonder why it is that your recordings are so afflicted by noise.  The self 
>> noise spec for the SPS200 is 12 dBA, which is similar to that of other 
>> soundfield microphones from Soundfield.  While 12 dBA isn't noise free, it 
>> should be pretty quiet.  As a reference, the average threshold of 
>> detectability for microphone noise is about 6 dBA, assuming a natural 
>> recording scenario.  That is, assuming that the sounds are replayed at the 
>> same level at which they occurred in the recording environment.
>> 
>> Of course, it may be that the microphone doesn't meet specifications.
>> 
>> I'm a bit confused by the recordings that you placed at
>> http://listen.ame.asu.edu/sonic_events.php
>> 
>> 
>> The first recording is labeled as "no audio".  The second recording is 
>> labeled as "you can hear Garth open his canteen and move some things 
>> around."  There's certainly a lot more noise in that second recording.  
>> About 46 dB more, unweighted.  It would be interesting to try to perform 
>> some more controlled recordings to find out whether the noise is coming from 
>> the mic, or not, and whether it meets specifications.
>> 
>> Do you ever get to the SF bay area?
>> 
>> Eric Benjamin
>> 
>> 
>> On Wednesday, August 6, 2014 3:12 PM, Sampo Syreeni  wrote:
>> 
>> 
>> 
>> On 2014-08-06, Joseph Anderson wrote:
>> 
>>> I take the noise profile from each individual A-format channel...
>> 
>> At the risk of sounding trite, what is noise? I'd argue that it isn't 
>> one thing, and that it's pretty difficult to define with mathematical 
>> precision. If you're talking about environmental background, then 
>> approaches like gating A-format or some other suitable directional 
>> representation of sound is a good idea.
>> 
>> If you're talking about tape noise instead, that isn't directional at 
>> all, at least until you get into directional masking calculations over 
>> what you can throw away without getting caught. In that case you'd want 
>> to operationalise what you consider noise, then find out an optimal way 
>> of extending that idea to B-format, and do the kind of joint processing 
>> Eero suggests.
>> 
>> The easiest way probably is to go with just W in the sidechain and equal 
>> gating for all the channels in the main one. The next step would be to 
>> do the same per frequency, and so on. However, in the ambisonic world, 
>> you'll then bump into a third source: the mic. Since the Soundfield 
>> works on differencing principles, W has a totally different noise 
>> profile from XYZ, and typically it only gets worse from there as the 
>> order goes up. (Or it doesn't; that depends wholly on the mic geometry.)
>> 
>> The point is, I don't think there is a monolithic thing called "noise" 
>> which can be just blindly "reduced". There never was even in monophonic 
>> recordings, and the free degrees of freedom in your signal chain just 
>> multiply when you go through stereo to ambisonic. So, you need to be 
>> careful about which source(s) of unwanted hiss, distortion or bogus 
>> sources you're talking about, you'll have to develop computationally 
>> tractable models of both your utility signal and the noise, and only 
>> then can you really start to combine all of the machinery into something 
>> which actually works/sounds good.
>> 
>> E.g. when you expand/limit A-format, implicitly your noise model is a 
>> hiss which is directional to first order and your model of the utility 
>> signal is something like a strong, wideband directional signal near it, 
>> which makes directional sine-to-noise masking statistics relevant. Break 
>> those conditions and bad things will most likely happen.
>> 
>> So, try your approach on a two sine test signal, separated in frequency 
>> more than a critical band's worth. Pan one of the sines due front, and 
>> revolve the other one around at about 1Hz and say -6dB. Then add pink 
>> noise at about -10dB to each of the B-format channels independently. I'm 
>> rather sure that while your approach will work beautifully for the front 
>> signal alone when adjusted right, it'll lead to nasty, anisotropic noise 
>> pumping with the dynamic signal in place.
>> 
>> (Oh, and by the way, which A-format? As long as you're dealing with a 
>> perfect mic and linear, time-invariant filteri

Re: [Sursound] Noise reduction on Ambisonic files

[Paul Hodges]

--On 07 August 2014 13:41 +0300 Eero Aro  wrote:

> Many Sursounders may not be aware that there are practically
> at all multichannel noise reduction systems available. They are all
> stereo, and can process mono.

Adobe Audition can do noise reduction on multi-channel files.

Paul

-- 
Paul Hodges

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Re: [Sursound] Noise reduction on Ambisonic files

[Eero Aro]

Hi All

Sampo, I guess that in this case we are talking about the hiss
that originates from the amplifier stages of the microphone.
Of course the acoustic noise into the capsules from Brownian
movement adds to this, but that part is much smaller.

The microphone self noise (or hiss, if you like) was the first
reason, why I decided to stop using the Soundfields (ST250 and
MK IV and V at that time) for radio drama. The hiss from the
mic was simply too audible for the scenes. Possibly the way that
the Finnish actors use their voice and dynamics is different to the
declamating yelling that I hear in radio drama from many other
countries. :-)

I ended up using the Soundfield only for background atmospheres
that I recorded separately.

Many Sursounders may not be aware that there are practically
at all multichannel noise reduction systems available. They are all
stereo, and can process mono. The same thing was true with
multichannel equalizers and dynamic processors at the time,
when I would have use of them in the nineties. Because 5.1 systems
became popular, there now are EQ:s and dynamic processors.
But no noise reduction systems.

Just one more note Garth; When you have denoised the files,
check that their length bitwise is the same as it used to be. At least
in the large scale you are on the safe side then. Of course the
phase may deviate during the file run, but I don't think that won't
happen.

Eero

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Re: [Sursound] Noise reduction on Ambisonic files

[David Worrall]

I never achieved satisfactory (electronically quiet) environmental recordings 
with the portable Soundfield. Maybe the ones I used weren't properly 
calibrated: they were certainly affected by humidity.
I have had much better luck with the TetraMic - as long as radio interference 
can be kept under control.
Garth: what cable lengths are you using and are the inputs balanced?

D.

On 07/08/2014, at 1:03 AM, Eric Benjamin wrote:

> Garth,
> 
> I wonder why it is that your recordings are so afflicted by noise.  The self 
> noise spec for the SPS200 is 12 dBA, which is similar to that of other 
> soundfield microphones from Soundfield.  While 12 dBA isn't noise free, it 
> should be pretty quiet.  As a reference, the average threshold of 
> detectability for microphone noise is about 6 dBA, assuming a natural 
> recording scenario.  That is, assuming that the sounds are replayed at the 
> same level at which they occurred in the recording environment.
> 
> Of course, it may be that the microphone doesn't meet specifications.
> 
> I'm a bit confused by the recordings that you placed at
> http://listen.ame.asu.edu/sonic_events.php
> 
> 
> The first recording is labeled as "no audio".  The second recording is 
> labeled as "you can hear Garth open his canteen and move some things around." 
>  There's certainly a lot more noise in that second recording.  About 46 dB 
> more, unweighted.  It would be interesting to try to perform some more 
> controlled recordings to find out whether the noise is coming from the mic, 
> or not, and whether it meets specifications.
> 
> Do you ever get to the SF bay area?
> 
> Eric Benjamin
> 
> 
> On Wednesday, August 6, 2014 3:12 PM, Sampo Syreeni  wrote:
> 
> 
> 
> On 2014-08-06, Joseph Anderson wrote:
> 
>> I take the noise profile from each individual A-format channel...
> 
> At the risk of sounding trite, what is noise? I'd argue that it isn't 
> one thing, and that it's pretty difficult to define with mathematical 
> precision. If you're talking about environmental background, then 
> approaches like gating A-format or some other suitable directional 
> representation of sound is a good idea.
> 
> If you're talking about tape noise instead, that isn't directional at 
> all, at least until you get into directional masking calculations over 
> what you can throw away without getting caught. In that case you'd want 
> to operationalise what you consider noise, then find out an optimal way 
> of extending that idea to B-format, and do the kind of joint processing 
> Eero suggests.
> 
> The easiest way probably is to go with just W in the sidechain and equal 
> gating for all the channels in the main one. The next step would be to 
> do the same per frequency, and so on. However, in the ambisonic world, 
> you'll then bump into a third source: the mic. Since the Soundfield 
> works on differencing principles, W has a totally different noise 
> profile from XYZ, and typically it only gets worse from there as the 
> order goes up. (Or it doesn't; that depends wholly on the mic geometry.)
> 
> The point is, I don't think there is a monolithic thing called "noise" 
> which can be just blindly "reduced". There never was even in monophonic 
> recordings, and the free degrees of freedom in your signal chain just 
> multiply when you go through stereo to ambisonic. So, you need to be 
> careful about which source(s) of unwanted hiss, distortion or bogus 
> sources you're talking about, you'll have to develop computationally 
> tractable models of both your utility signal and the noise, and only 
> then can you really start to combine all of the machinery into something 
> which actually works/sounds good.
> 
> E.g. when you expand/limit A-format, implicitly your noise model is a 
> hiss which is directional to first order and your model of the utility 
> signal is something like a strong, wideband directional signal near it, 
> which makes directional sine-to-noise masking statistics relevant. Break 
> those conditions and bad things will most likely happen.
> 
> So, try your approach on a two sine test signal, separated in frequency 
> more than a critical band's worth. Pan one of the sines due front, and 
> revolve the other one around at about 1Hz and say -6dB. Then add pink 
> noise at about -10dB to each of the B-format channels independently. I'm 
> rather sure that while your approach will work beautifully for the front 
> signal alone when adjusted right, it'll lead to nasty, anisotropic noise 
> pumping with the dynamic signal in place.
> 
> (Oh, and by the way, which A-format? As long as you're dealing with a 
> perfect mic and linear, time-invariant filtering operation, you don't 
> have to think about that because you can go willy nilly between A and B. 
> But once you start applying this kind of processing, every possible 
> orientation of the mic gives rise to a separate A-format. Which one 
> should it be? The above example presumes one of the capsules is facing 
> towards 

Re: [Sursound] Noise reduction on Ambisonic files

[Garth Paine]

Thanks Sampo - yes I am aware of these issues, but in this case simply have the 
SPS200 connected direct to the Sound Devices 788T, so the path is very simple.

Cheers, Garth


On Aug 6, 2014, at 5:19 PM, Sampo Syreeni  wrote:

> On 2014-08-06, Garth Paine wrote:
> 
>> I have always found the SPS200 and even the older Soundfield mics noisy and 
>> find the Core mic unusable in ambient environments - of course in a music 
>> concert it is a different story as the program amplitude is generally much 
>> higher.
> 
> Might be a stupid question, but... Have you ever truly quantified the gain 
> structure of your whole rig, from start to end, piece by piece? In absolute 
> amplitude, weighted or not?
> 
> I mean while you can't "exactly" call it a rookie mistake, it's still pretty 
> common that people just leave too much headroom, and in all the wrong places, 
> and don't even come to think of evening it out over the whole chain.
> 
> That's especially common with the kinds of big, high voltage, old time, pro 
> mics like the original Soundfield, because they're really rated to meet their 
> specs at the very edge of their envelope. If they say it takes 120dB, it 
> takes it; don't be squeamish. You don't really leave any headroom there, even 
> if you plan on actually, physically going to 120dB for the duration. And for 
> the most part, you won't actually go anywhere near that, so that you can 
> safely turn the thing down, bringing the noise down along with it.
> 
> Then, when you've set the maximum amplitude at the input stage, you always 
> work down from there. It ain't gonna get louder unless you made it so, and in 
> that case, you should recalibrate the stage in the chain so that it doesn't 
> amplify, but at most once reaches the maximum amplitude it can take in a 
> fully linear fashion.
> 
> In live, ambient situations you might be tempted to leave headroom just in 
> case. But seriously, what's the probability of a sonic boom happening right 
> above, or the environment otherwise changing so fast you can't gain-ride it? 
> Pretty much zilch. Or if it's not, well then you ought to be running multiple 
> mics side by side at different gains, kind of like they do HDR photography.
> 
> I know, this is recording-fu 101 or something. But at the same time it's 
> still the reason for the vast majority of noise problems. Especially when 
> dealing with exotic mics such as soundfields, at the crucial first stage of a 
> recording.
> -- 
> Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front
> +358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2
> ___
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Re: [Sursound] Noise reduction on Ambisonic files

[Sampo Syreeni]

On 2014-08-06, Garth Paine wrote:

I have always found the SPS200 and even the older Soundfield mics 
noisy and find the Core mic unusable in ambient environments - of 
course in a music concert it is a different story as the program 
amplitude is generally much higher.


Might be a stupid question, but... Have you ever truly quantified the 
gain structure of your whole rig, from start to end, piece by piece? In 
absolute amplitude, weighted or not?


I mean while you can't "exactly" call it a rookie mistake, it's still 
pretty common that people just leave too much headroom, and in all the 
wrong places, and don't even come to think of evening it out over the 
whole chain.


That's especially common with the kinds of big, high voltage, old time, 
pro mics like the original Soundfield, because they're really rated to 
meet their specs at the very edge of their envelope. If they say it 
takes 120dB, it takes it; don't be squeamish. You don't really leave any 
headroom there, even if you plan on actually, physically going to 120dB 
for the duration. And for the most part, you won't actually go anywhere 
near that, so that you can safely turn the thing down, bringing the 
noise down along with it.


Then, when you've set the maximum amplitude at the input stage, you 
always work down from there. It ain't gonna get louder unless you made 
it so, and in that case, you should recalibrate the stage in the chain 
so that it doesn't amplify, but at most once reaches the maximum 
amplitude it can take in a fully linear fashion.


In live, ambient situations you might be tempted to leave headroom just 
in case. But seriously, what's the probability of a sonic boom happening 
right above, or the environment otherwise changing so fast you can't 
gain-ride it? Pretty much zilch. Or if it's not, well then you ought to 
be running multiple mics side by side at different gains, kind of like 
they do HDR photography.


I know, this is recording-fu 101 or something. But at the same time it's 
still the reason for the vast majority of noise problems. Especially 
when dealing with exotic mics such as soundfields, at the crucial first 
stage of a recording.

--
Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front
+358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2
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Re: [Sursound] Noise reduction on Ambisonic files

[Garth Paine]

Hi Sampo

Yes your philosophical meanderings are indeed some of my concerns about image - 
I am not convinced that in rendering a processed B-Format file that it would 
decode well in all output formats - Binaural, Stereo, 5.1 etc and I am not a 
terribly technical DSP person so running experiments to accurately check the 
phase is beyond me.  AS you mention, the W is a separate issue and I have 
thought as Joseph argues of taking a snapshot of the background noise at a 
quiet point (although all the recordings are quiet) for each capsule and then 
applying them in A-Format (tracks separated) before the B-Format conversion. In 
this case I am doing that only for the SPS200 for which I trust Soundfield 
provided me with the right decoding in their software, although I do often feel 
it is a few degrees off to the right, but thats another story.

I guess now that others have suggested it works for them I will apply RX to the 
A-Format tracks and see what happens.  It does seem strange that there is not a 
commercially available system (that I can afford of course - i.e.. not a System 
6000 solution) that does this automatically and guarantees the phase.

Cheers, Garth

On Aug 6, 2014, at 3:12 PM, Sampo Syreeni  wrote:

> On 2014-08-06, Joseph Anderson wrote:
> 
>> I take the noise profile from each individual A-format channel...
> 
> At the risk of sounding trite, what is noise? I'd argue that it isn't one 
> thing, and that it's pretty difficult to define with mathematical precision. 
> If you're talking about environmental background, then approaches like gating 
> A-format or some other suitable directional representation of sound is a good 
> idea.
> 
> If you're talking about tape noise instead, that isn't directional at all, at 
> least until you get into directional masking calculations over what you can 
> throw away without getting caught. In that case you'd want to operationalise 
> what you consider noise, then find out an optimal way of extending that idea 
> to B-format, and do the kind of joint processing Eero suggests.
> 
> The easiest way probably is to go with just W in the sidechain and equal 
> gating for all the channels in the main one. The next step would be to do the 
> same per frequency, and so on. However, in the ambisonic world, you'll then 
> bump into a third source: the mic. Since the Soundfield works on differencing 
> principles, W has a totally different noise profile from XYZ, and typically 
> it only gets worse from there as the order goes up. (Or it doesn't; that 
> depends wholly on the mic geometry.)
> 
> The point is, I don't think there is a monolithic thing called "noise" which 
> can be just blindly "reduced". There never was even in monophonic recordings, 
> and the free degrees of freedom in your signal chain just multiply when you 
> go through stereo to ambisonic. So, you need to be careful about which 
> source(s) of unwanted hiss, distortion or bogus sources you're talking about, 
> you'll have to develop computationally tractable models of both your utility 
> signal and the noise, and only then can you really start to combine all of 
> the machinery into something which actually works/sounds good.
> 
> E.g. when you expand/limit A-format, implicitly your noise model is a hiss 
> which is directional to first order and your model of the utility signal is 
> something like a strong, wideband directional signal near it, which makes 
> directional sine-to-noise masking statistics relevant. Break those conditions 
> and bad things will most likely happen.
> 
> So, try your approach on a two sine test signal, separated in frequency more 
> than a critical band's worth. Pan one of the sines due front, and revolve the 
> other one around at about 1Hz and say -6dB. Then add pink noise at about 
> -10dB to each of the B-format channels independently. I'm rather sure that 
> while your approach will work beautifully for the front signal alone when 
> adjusted right, it'll lead to nasty, anisotropic noise pumping with the 
> dynamic signal in place.
> 
> (Oh, and by the way, which A-format? As long as you're dealing with a perfect 
> mic and linear, time-invariant filtering operation, you don't have to think 
> about that because you can go willy nilly between A and B. But once you start 
> applying this kind of processing, every possible orientation of the mic gives 
> rise to a separate A-format. Which one should it be? The above example 
> presumes one of the capsules is facing towards the reference. It gets much 
> worse if you place the source directly between three adjacent capsules, in 
> angle space.)
> -- 
> Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front
> +358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2
> ___
> 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.

Re: [Sursound] Noise reduction on Ambisonic files

[Garth Paine]

HI Eric

This has been an interesting discussion already - thanks to everyone for their 
input.

I think the students I employed to tagg the recordings took great pleasure in 
putting me down for all the extraneous noises they hear!  

If you got to the Listen site http://listen.ame.asu.edu/sonic_events.php and 
type in Goose in the TAG box in the search it will bring up an entry that if 
you click will then play in the bar above the search.  This is a much better 
example.  I also use Schoeps M/S pair on another 2 channels of the 788 and find 
them much much cleaner even with higher recording gain.  I have always found 
the SPS200 and even the older Soundfield mics noisy and find the Core mic 
unusable in ambient environments - of course in a music concert it is a 
different story as the program amplitude is generally much higher.

Yes indeed I can get over the SF.  I am in Phoenix and love the sea so any 
opportunity ;-)



On Aug 6, 2014, at 4:03 PM, Eric Benjamin  wrote:

> Garth,
> 
> I wonder why it is that your recordings are so afflicted by noise.  The self 
> noise spec for the SPS200 is 12 dBA, which is similar to that of other 
> soundfield microphones from Soundfield.  While 12 dBA isn't noise free, it 
> should be pretty quiet.  As a reference, the average threshold of 
> detectability for microphone noise is about 6 dBA, assuming a natural 
> recording scenario.  That is, assuming that the sounds are replayed at the 
> same level at which they occurred in the recording environment.
> 
> Of course, it may be that the microphone doesn't meet specifications.
> 
> I'm a bit confused by the recordings that you placed at
> http://listen.ame.asu.edu/sonic_events.php
> 
> 
> The first recording is labeled as "no audio".  The second recording is 
> labeled as "you can hear Garth open his canteen and move some things around." 
>  There's certainly a lot more noise in that second recording.  About 46 dB 
> more, unweighted.  It would be interesting to try to perform some more 
> controlled recordings to find out whether the noise is coming from the mic, 
> or not, and whether it meets specifications.
> 
> Do you ever get to the SF bay area?
> 
> Eric Benjamin
> 
> 
> On Wednesday, August 6, 2014 3:12 PM, Sampo Syreeni  wrote:
> 
> 
> 
> On 2014-08-06, Joseph Anderson wrote:
> 
>> I take the noise profile from each individual A-format channel...
> 
> At the risk of sounding trite, what is noise? I'd argue that it isn't 
> one thing, and that it's pretty difficult to define with mathematical 
> precision. If you're talking about environmental background, then 
> approaches like gating A-format or some other suitable directional 
> representation of sound is a good idea.
> 
> If you're talking about tape noise instead, that isn't directional at 
> all, at least until you get into directional masking calculations over 
> what you can throw away without getting caught. In that case you'd want 
> to operationalise what you consider noise, then find out an optimal way 
> of extending that idea to B-format, and do the kind of joint processing 
> Eero suggests.
> 
> The easiest way probably is to go with just W in the sidechain and equal 
> gating for all the channels in the main one. The next step would be to 
> do the same per frequency, and so on. However, in the ambisonic world, 
> you'll then bump into a third source: the mic. Since the Soundfield 
> works on differencing principles, W has a totally different noise 
> profile from XYZ, and typically it only gets worse from there as the 
> order goes up. (Or it doesn't; that depends wholly on the mic geometry.)
> 
> The point is, I don't think there is a monolithic thing called "noise" 
> which can be just blindly "reduced". There never was even in monophonic 
> recordings, and the free degrees of freedom in your signal chain just 
> multiply when you go through stereo to ambisonic. So, you need to be 
> careful about which source(s) of unwanted hiss, distortion or bogus 
> sources you're talking about, you'll have to develop computationally 
> tractable models of both your utility signal and the noise, and only 
> then can you really start to combine all of the machinery into something 
> which actually works/sounds good.
> 
> E.g. when you expand/limit A-format, implicitly your noise model is a 
> hiss which is directional to first order and your model of the utility 
> signal is something like a strong, wideband directional signal near it, 
> which makes directional sine-to-noise masking statistics relevant. Break 
> those conditions and bad things will most likely happen.
> 
> So, try your approach on a two sine test signal, separated in frequency 
> more than a critical band's worth. Pan one of the sines due front, and 
> revolve the other one around at about 1Hz and say -6dB. Then add pink 
> noise at about -10dB to each of the B-format channels independently. I'm 
> rather sure that while your approach will work beautifully for the front 
> signal alo

Re: [Sursound] Noise reduction on Ambisonic files

[Eric Benjamin]

Garth,

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

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

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


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

Do you ever get to the SF bay area?

Eric Benjamin


On Wednesday, August 6, 2014 3:12 PM, Sampo Syreeni  wrote:
 


On 2014-08-06, Joseph Anderson wrote:

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

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

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

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

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

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

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

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

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Re: [Sursound] Noise reduction on Ambisonic files

[Sampo Syreeni]

On 2014-08-06, Joseph Anderson wrote:


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


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


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


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


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


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


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


(Oh, and by the way, which A-format? As long as you're dealing with a 
perfect mic and linear, time-invariant filtering operation, you don't 
have to think about that because you can go willy nilly between A and B. 
But once you start applying this kind of processing, every possible 
orientation of the mic gives rise to a separate A-format. Which one 
should it be? The above example presumes one of the capsules is facing 
towards the reference. It gets much worse if you place the source 
directly between three adjacent capsules, in angle space.)

--
Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front
+358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2
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Re: [Sursound] Noise reduction on Ambisonic files

[Joseph Anderson]

Hello Eero,

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





Joseph Anderson

j.ander...@ambisonictoolkit.net
http://www.ambisonictoolkit.net




On 6 Aug 2014, at 1:04 pm, Eero Aro  wrote:

> Joseph Anderson wrote
> 
>> In B-format, if you gate out one of the channels, you get a substantial 
>> change in imaging. E.g., if you gate out Y, you loose 'width'.
> 
> This makes me think, that with Izotope RX or similar noise reduction software,
> it might be good to take the noise profile sample for example from W and X
> channels, run the noise reduction and then use the same profile for the
> Y and Z channels. Don't take a new noise profile from Y and Z.
> 
> Or - if you prefer for some reason running the noise reduction for all 
> B-format
> signals separately, use the noise profile from just one (for example W) for
> all of the channels.
> 
> Eero
> 
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Re: [Sursound] Noise reduction on Ambisonic files

[Eero Aro]

Joseph Anderson wrote

In B-format, if you gate out one of the channels, you get a substantial change in imaging. E.g., if you 
gate out Y, you loose 'width'.


This makes me think, that with Izotope RX or similar noise reduction software,
it might be good to take the noise profile sample for example from W and X
channels, run the noise reduction and then use the same profile for the
Y and Z channels. Don't take a new noise profile from Y and Z.

Or - if you prefer for some reason running the noise reduction for all B-format
signals separately, use the noise profile from just one (for example W) for
all of the channels.

Eero

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Re: [Sursound] Noise reduction on Ambisonic files

[Joseph Anderson]

Hello David,

You can think about it intuitively...

Broadband de-noising is a frequency dependent gating and/or gain reduction 
process.

In B-format, if you gate out one of the channels, you get a substantial change 
in imaging. E.g., if you gate out Y, you loose 'width'.

In A-format, you're gating out a segment (I think in terms of pie slices) of 
the soundfield. E.g., gate out Front-Left-Up. Here, you also have imaging 
distortion effects, but, they're similar to what happens in stereo. E.g., gate 
out Left.

Make sense?

Now... there may be an argument as to whether you want any image distortion to 
happen... de-noising is about throwing (or suppressing) stuff away.


My best,



Joseph Anderson

j.ander...@ambisonictoolkit.net
http://www.ambisonictoolkit.net




On 5 Aug 2014, at 3:52 pm, David Pickett  wrote:

> At 20:56 05-08-14, Joseph Anderson wrote:
> 
> >I'd advise converting from B-format to A-format, then doing all
> >de-noising, compression, etc, in A-format. Followed by, re-encoding
> >back to B-format.
> 
> What's the theory that predicts that the results will be any different than 
> doing it on B-format, given that the transform is a linear matrix?
> 
> David
> 
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Re: [Sursound] Noise reduction on Ambisonic files

[David Pickett]

At 20:56 05-08-14, Joseph Anderson wrote:

>I'd advise converting from B-format to A-format, then doing all
>de-noising, compression, etc, in A-format. Followed by, re-encoding
>back to B-format.

What's the theory that predicts that the results will be any 
different than doing it on B-format, given that the transform is a 
linear matrix?


David

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Re: [Sursound] Noise reduction on Ambisonic files

[Garth Paine]

Hi Dave

Nice to hear from you and thanks for your input - it seems strange to me, given 
the known self noise of the most prevalent ambisonic microphones that there is 
not a solution out there already.  Indeed your summations of the process aligns 
with mine, but I am somewhat nervous about looking to apply this over all 4 
channels as there is so little across the 4 channels that I could use as a 
common measure of phase accuracy after processing, and to be honest I am not 
looking to write code for this as DSP is not my strong point - but I would see 
a use for this across the community.

I would of course be happy to apply the noise reduction to the B-Format file.  
The idea for the Listen(n) project is to provide a wind range of listening 
outcomes from mobile devices with headphones to surround sound setups - so the 
decoding would need to be simple and be applicable across domestic platforms - 
so I am imagining that the noise reduction would therefore need to happen pre 
decoding to the listening format?

Would love to find a solution - It has been suggested for instance that I use 
single instances of Izotopes RX on each of the 4 channels for the A-Format file 
and load a noise template in each - still I am concerned about any phase 
variation pre to decoding.  Am I being over concerned?

Cheers, Garth 


On Aug 5, 2014, at 4:09 AM, Dave Malham  wrote:

> Hi Garth,
>   An interesting one. certainly got me thinking - trouble is, you don't
> really want thoughts but measurements. I suspect it depends a lot on what
> the internal mechanism of the noise reduction system is. Mostly, as far as
> I can ascertain, there's an analysis filter bank to split the sound into
> bands which are then subject to some sort of processing, then the bands are
> re-combined somehow either directly or by resynthesis to produce the
> output. The most critical thing will usually be the combination of the
> analysis and resynthesis  steps. For instance, a well designed and well
> implemented FFT/iFFT pair should preserve the phase well. However, since
> you rarely have access to the internals of these things for analysis,
> measurement - or just listening with a good pair of ears - is the only way
> forward.
> 
> I suspect that processing the B format after conversion from A would be the
> best - anyone else have any thoughts?
> 
> Dave
> 
> PS Of course, you could just always process the speaker feeds, for know, as
> that would be the least risky  but most processing heavy option
> 
> 
> On 4 August 2014 20:23, Garth Paine  wrote:
> 
>> Hi everyone
>> 
>> I have been doing a lot of ambient Ambisonic A format recordings (sps200
>> into SD788) and as the environmental levels are so low the self noise of
>> the microphone becomes a bit of an issue on playback - I have RX for stereo
>> noise reduction but have not found a solution for multichannel that would
>> make me relaxed about maintaining the phase for decoding - I want to output
>> B-Format so decoding onto any speaker array rather than just output 5.1 and
>> use a surround noise cleaner.  I would appreciate thoughts from the list -
>> I am guessing as the Soundfield mics are know for self noise that others
>> have faced and perhaps solved this issue already?  thanks in advance
>> 
>> ps. you can hear some of the recordings here
>> http://listen.ame.asu.edu/sonic_events.php
>> 
>> Cheers,
>> Garth Paine
>> ga...@activatedspace.com
>> 
>> 
>> -- next part --
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>> 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] Noise reduction on Ambisonic files

[Joseph Anderson]

Hello Garth, Dave,

I'd advise converting from B-format to A-format, then doing all de-noising, 
compression, etc, in A-format. Followed by, re-encoding back to B-format.

This is how I've done all my work (both acousmatic involving field-recordings 
and musical location recordings). Depending on your de-noiser / gating 
settings, you can get image distortion effects... but these are similar to 
working with stereo.

If you really want to be precious, instead of just 4 channels of A-format (for 
FOA), you can decode to more (say 6 channels as vertices of an octahedron, or 8 
channels, in a cube), do your processing and then re-encode back to B-format.

Also, you can adjust the polar patterns of your decode. Using 
'controlled-opposites' decode can give you a smoother result, depending on your 
material. If you use a more custom version of A-format (octahedron or cube), 
make sure your re-encoding matrix is correctly scaled, so you get the correct 
balance of W vs X,Y,Z on re-encoding.

Hope this helps!


My best,


Joseph Anderson

j.ander...@ambisonictoolkit.net
http://www.ambisonictoolkit.net




On 5 Aug 2014, at 4:09 am, Dave Malham  wrote:

> Hi Garth,
>   An interesting one. certainly got me thinking - trouble is, you don't
> really want thoughts but measurements. I suspect it depends a lot on what
> the internal mechanism of the noise reduction system is. Mostly, as far as
> I can ascertain, there's an analysis filter bank to split the sound into
> bands which are then subject to some sort of processing, then the bands are
> re-combined somehow either directly or by resynthesis to produce the
> output. The most critical thing will usually be the combination of the
> analysis and resynthesis  steps. For instance, a well designed and well
> implemented FFT/iFFT pair should preserve the phase well. However, since
> you rarely have access to the internals of these things for analysis,
> measurement - or just listening with a good pair of ears - is the only way
> forward.
> 
> I suspect that processing the B format after conversion from A would be the
> best - anyone else have any thoughts?
> 
> Dave
> 
> PS Of course, you could just always process the speaker feeds, for know, as
> that would be the least risky  but most processing heavy option
> 
> 
> On 4 August 2014 20:23, Garth Paine  wrote:
> 
>> Hi everyone
>> 
>> I have been doing a lot of ambient Ambisonic A format recordings (sps200
>> into SD788) and as the environmental levels are so low the self noise of
>> the microphone becomes a bit of an issue on playback - I have RX for stereo
>> noise reduction but have not found a solution for multichannel that would
>> make me relaxed about maintaining the phase for decoding - I want to output
>> B-Format so decoding onto any speaker array rather than just output 5.1 and
>> use a surround noise cleaner.  I would appreciate thoughts from the list -
>> I am guessing as the Soundfield mics are know for self noise that others
>> have faced and perhaps solved this issue already?  thanks in advance
>> 
>> ps. you can hear some of the recordings here
>> http://listen.ame.asu.edu/sonic_events.php
>> 
>> Cheers,
>> Garth Paine
>> ga...@activatedspace.com
>> 
>> 
>> -- next part --
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>> https://mail.music.vt.edu/mailman/private/sursound/attachments/20140804/ca2c4e9f/attachment.html
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>> ___
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>> 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] Noise reduction on Ambisonic files

[Paul Hodges]

--On 05 August 2014 12:09 +0100 Dave Malham 
wrote:

> I suspect that processing the B format after conversion from A would
> be the best - anyone else have any thoughts?

I have used RX on two-channel pairs of the B-format, with no obvious
breaking of the reconstruction.

Last time I had a noise problem, it was a single capsule, so I
processed that channel of the A-format alone (using RX) before making
the B-format, and the results seemed OK.

If you want to process four channels together, Audition can be
persuaded to open and rewrite a four-channel file correctly (even
though it can't create one).

Paul

-- 
Paul Hodges

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Re: [Sursound] Noise reduction on Ambisonic files

[Dave Malham]

Hi Garth,
   An interesting one. certainly got me thinking - trouble is, you don't
really want thoughts but measurements. I suspect it depends a lot on what
the internal mechanism of the noise reduction system is. Mostly, as far as
I can ascertain, there's an analysis filter bank to split the sound into
bands which are then subject to some sort of processing, then the bands are
re-combined somehow either directly or by resynthesis to produce the
output. The most critical thing will usually be the combination of the
analysis and resynthesis  steps. For instance, a well designed and well
implemented FFT/iFFT pair should preserve the phase well. However, since
you rarely have access to the internals of these things for analysis,
measurement - or just listening with a good pair of ears - is the only way
forward.

I suspect that processing the B format after conversion from A would be the
best - anyone else have any thoughts?

 Dave

PS Of course, you could just always process the speaker feeds, for know, as
that would be the least risky  but most processing heavy option


On 4 August 2014 20:23, Garth Paine  wrote:

> Hi everyone
>
> I have been doing a lot of ambient Ambisonic A format recordings (sps200
> into SD788) and as the environmental levels are so low the self noise of
> the microphone becomes a bit of an issue on playback - I have RX for stereo
> noise reduction but have not found a solution for multichannel that would
> make me relaxed about maintaining the phase for decoding - I want to output
> B-Format so decoding onto any speaker array rather than just output 5.1 and
> use a surround noise cleaner.  I would appreciate thoughts from the list -
> I am guessing as the Soundfield mics are know for self noise that others
> have faced and perhaps solved this issue already?  thanks in advance
>
> ps. you can hear some of the recordings here
> http://listen.ame.asu.edu/sonic_events.php
>
> Cheers,
> Garth Paine
> ga...@activatedspace.com
>
>
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>



-- 

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] Noise reduction on Ambisonic files

[Garth Paine]

Hi everyone

I have been doing a lot of ambient Ambisonic A format recordings (sps200 into 
SD788) and as the environmental levels are so low the self noise of the 
microphone becomes a bit of an issue on playback - I have RX for stereo noise 
reduction but have not found a solution for multichannel that would make me 
relaxed about maintaining the phase for decoding - I want to output B-Format so 
decoding onto any speaker array rather than just output 5.1 and use a surround 
noise cleaner.  I would appreciate thoughts from the list - I am guessing as 
the Soundfield mics are know for self noise that others have faced and perhaps 
solved this issue already?  thanks in advance 

ps. you can hear some of the recordings here 
http://listen.ame.asu.edu/sonic_events.php 

Cheers,
Garth Paine
ga...@activatedspace.com


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