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[12345]

No, but after reading about his gravity generator, I am reminded of a research project at Rensselaer Polytechnic using microwaves to reduce drag and heat transfer, and aid propulsion.  

http://www.abovetopsecret.com/pages/airspike.html

Cool!

MW

[dcollins]

My World wrote on Mon, 17 October 2005 22:01

No, but after reading about his gravity generator, I am reminded of a research project at Rensselaer Polytechnic using microwaves to reduce drag and heat transfer, and aid propulsion.   http://www.abovetopsecret.com/pages/airspike.html Cool!

C'mon, admit it!

DC

[12345]

A good link on the Berkeley smart dust:
http://www-bsac.eecs.berkeley.edu/archive/users/warneke-bret t/SmartDust/

[12345]

dcollins wrote on Mon, 17 October 2005 22:12

My World wrote on Mon, 17 October 2005 22:01 No, but after reading about his gravity generator, I am reminded of a research project at Rensselaer Polytechnic using microwaves to reduce drag and heat transfer, and aid propulsion.   http://www.abovetopsecret.com/pages/airspike.html Cool! C'mon, admit it! DC

That is some funny stuff, DC

MW

[dcollins]

My World wrote on Mon, 17 October 2005 22:19

A good link on the Berkeley smart dust:  http://www-bsac.eecs.berkeley.edu/archive/users/warneke-bret t/SmartDust/

http://www.technologynewsdaily.com/node/1538

Another form of smart dust.............

DC

[Ronny]

dcollins wrote on Tue, 18 October 2005 01:27

My World wrote on Mon, 17 October 2005 22:19 A good link on the Berkeley smart dust:   http://www-bsac.eecs.berkeley.edu/archive/users/warneke-bret t/SmartDust/ http://www.technologynewsdaily.com/node/1538 Another form of smart dust............. DC

More like parasites than dust.

[mandel]

To sum (correct me if I am wrong)...

if we record and store at 24/48, the data playback should theoretically be indistinguishable from 24/192 or 24/384, using Nyquist and Fourier laws...more info does not mean more accuracy or resolution here, since Fourier analysis says all one needs is frequencies up to the Nyquist frequency to break down all audible signals into integral of sine curves...The only difficulty, and the reason why digital can sound bad, is in the implementation of filters and other errors...if "perfect" filters are used and quantization errors are neglible, then sound reproduction should be "perfect" - no different from that in a "live" stage...

am I right here?

so, the question...I'm not challenging Nyquist and Fourier, but Fourier analysis is after all mathematical theorem, not a physics one...so could it be possible that using wavelet technology as complement, sound reproduction could be improved...better filters or easier filters to implement, and/or since wavelet does not deal with sampling rate, could there be something outside sampling rate that one could measure and could better represent the ear-brain thingy?

or would you argue that perfect is already perfect?

[Jon Hodgson]

mandel wrote on Tue, 18 October 2005 09:38

To sum (correct me if I am wrong)... if we record and store at 24/48, the data playback should theoretically be indistinguishable from 24/192 or 24/384, using Nyquist and Fourier laws...more info does not mean more accuracy or resolution here, since Fourier analysis says all one needs is frequencies up to the Nyquist frequency to break down all audible signals into integral of sine curves...The only difficulty, and the reason why digital can sound bad, is in the implementation of filters and other errors...if "perfect" filters are used and quantization errors are neglible, then sound reproduction should be "perfect" - no different from that in a "live" stage... am I right here?

Well the system is only going to be as good as the mics and speakers, but you get the general idea.

mandel wrote on Tue, 18 October 2005 09:38

so, the question...I'm not challenging Nyquist and Fourier, but Fourier analysis is after all mathematical theorem, not a physics one...so could it be possible that using wavelet technology as complement, sound reproduction could be improved...better filters or easier filters to implement, and/or since wavelet does not deal with sampling rate, could there be something outside sampling rate that one could measure and could better represent the ear-brain thingy? or would you argue that perfect is already perfect?

Filters and Fourier work very well together, so I doubt wavelets would make life any easier there.

It is possible that a model which more closely matched the ear/brain response, in both its strengths and weaknesses, would allow a better capture system, however I don't find it very likely because

1) The system would inevitably be more complex than a constant rate sampling system, complexity tends to create more problems.

2) We can already sample what most research tells us is more than the neccessary bandwidth with noise and distortion levels lower than the other links in the chain presently achieve, and perhaps will ever achieve.

So barring some radical shakeup of our knowledge of audio perception, I would say that constant rate sampling will probably remain the best option as an interface between analogue and digital.

IMHO A wavelet based model is more likely to be of use in lossy compression, signal analysiz and processing, and sound synthesis. So basically things you do to the PCM stream in between sampling it and playing it back.

[mandel]

Jon Hodgson wrote on Tue, 18 October 2005 12:29

IMHO A wavelet based model is more likely to be of use in lossy compression, signal analysiz and processing, and sound synthesis. So basically things you do to the PCM stream in between sampling it and playing it back.

Signal analysis and processing...would this make a difference to sound reproduction ie. more transparent sound produced?

(since we've established no higher sampling rate or data is required, at least not with our present knowledge of audio physics...)

Im asking lots of questions...

[maxdimario]

Quote:

Sennheiser RF microphones sort of work on this concept, using radio frequencies to measure the deflection of a microphone diaphragm.

they use RF to lower the impedance of the output coming out of the capsule and get a good output level in relation to that impedance.

in standard condensers a DC voltage is applied, and as the capsule membrane gets closer or more distant from the backplate the change in capacitance generates a voltage which is extremely low in current, hence the need for tubes and fets and 400 Meg resistors etc.

the sennheiser mic uses a high frequency that passes through the capsule.

as you might know the higher the frequency the lower the resistance of a capacitor.

so they use the rf to go through the capsule (which modulates the RF signal), then rectify it to get the analog waveform of the capsule movement.

sort of like decoding an AM radio signal, but lower frequency.

nothing like he was talking about I'm sure.

[Jon Hodgson]

mandel wrote on Tue, 18 October 2005 13:15

Jon Hodgson wrote on Tue, 18 October 2005 12:29 IMHO A wavelet based model is more likely to be of use in lossy compression, signal analysiz and processing, and sound synthesis. So basically things you do to the PCM stream in between sampling it and playing it back. Signal analysis and processing...would this make a difference to sound reproduction ie. more transparent sound produced? (since we've established no higher sampling rate or data is required, at least not with our present knowledge of audio physics...) Im asking lots of questions...

Well combining knowledge of the sound with knowledge of the speakers and possibly of the microphones, might possibly allow you to process the signal in such a way as to reduce the perceived distortions created by them, and thus make the sound more natural.

As a far simpler example of what I mean, BBE claim that part of their processing, by changing the phase relationship of high and low frequencies, cancels out the deficiencies of speakers in this respect and the sound that reaches your ears is more natural as a result.

[lord]

Can't say I've seen Shrek 4D. But I have been through the Disney Imagineering campus where they come up with all those rides, and spyed some pretty mind-blowing stuff. A lot of interesting audio experimentation going on in there, especially in reproduction.

Fascinating.

I want everyone to keep taking their meds, mmm-kay?

[dcollins]

Jon Hodgson wrote on Tue, 18 October 2005 06:10

BBE claim that part of their processing, by changing the phase relationship of high and low frequencies, cancels out the deficiencies of speakers in this respect and the sound that reaches your ears is more natural as a result.

How do they know what kind of speakers I use?
Or is there a DIP switch?

DC

[Jon Hodgson]

dcollins wrote on Thu, 20 October 2005 21:17

Jon Hodgson wrote on Tue, 18 October 2005 06:10 BBE claim that part of their processing, by changing the phase relationship of high and low frequencies, cancels out the deficiencies of speakers in this respect and the sound that reaches your ears is more natural as a result. How do they know what kind of speakers I use? Or is there a DIP switch? DC

They've been peeking...

What's more disturbing is they also told me what colour pyjamas you wear.  

But you're quite right, even if they're right about the reason for what you perceive, they can only do it according to some arbitrary reference... what they judge to be the average case perhaps.

Personally I think that compensation for speakers, if it's going to happen, should really be happening in the playback system.

[Ronny]

Jon Hodgson wrote on Thu, 20 October 2005 16:28

dcollins wrote on Thu, 20 October 2005 21:17 Jon Hodgson wrote on Tue, 18 October 2005 06:10 BBE claim that part of their processing, by changing the phase relationship of high and low frequencies, cancels out the deficiencies of speakers in this respect and the sound that reaches your ears is more natural as a result. How do they know what kind of speakers I use? Or is there a DIP switch? DC They've been peeking... What's more disturbing is they also told me what colour pyjamas you wear.   But you're quite right, even if they're right about the reason for what you perceive, they can only do it according to some arbitrary reference... what they judge to be the average case perhaps. Personally I think that compensation for speakers, if it's going to happen, should really be happening in the playback system.

The only real way is to tri-amp and delay the speaker lines relative to the distance between the horns, mids and bottom end.

I evaluated the BBE for Thorobred when it first came out, must have been 12 years or so ago. I mirrored the sound by eq'ing 6 to 16k with a shelf boosted by +5dB and used a parametric on the low end, boosting, IIRC +3dB with a bell curve around 100Hz. Had several people listen in the blind while I toggled between the BBE and the eq'd no BBE path and they couldn't pick the two out. My conclusion is that the BBE mainly raises gain and relies on the louder is better phenomenon to sell the device.