Wide bandwidth = necessary?

First of all I would like to thank all of you for the wonderful comments. The reason for this post was to get clarification from you the need for the ability to reproduce frequencies beyond human hearing.
In the past some manufacturers like Pioneer tried to "reconstruct" the signal content above 20kHz by using special filters like the "Legato Link". This was done because the designers believed that the frequencies above the 20kHz (up to 40kHz) were necessary for a natural sound. This is also one of the arguments of the vinyl people. But if the mics are capable only of picking up frequencies up to 20kHz, why the need for gear that can reproduce frequencies above that (if that is already bandlimited in the first place)? In case of vinyl: I truly believe cartridges are capable of reproducing frequencies above 20kHz, but if the vinyl itself contains no frequencies above let's say 16kHz at all, what is the use for such a bandwidth?

Chris

Well Dazzdax,I can refer you to an old Hi-Fi review (1982) of 5 cartridges by Martin Colloms who was in the lab testing those cartrides square wave responses out to 40Khz for his review,this is in the members section articles at The Vinyl Engine.I at one time hooked up my phono stage to my soundcard and using the old Cool Edit program recorded some Led Zepplin at 96/24 and there was ultrasonic content definitely present out to 40Khz.This was actually to duplicate an experiment by John Atkinson of Stereophile who did an article about the high resolution of vinyl (still in the archives).The Sheffield Direct to Disk series used to advertise how their recordings captured sonic information out to 50Khz.

Amazing! Thank you for the references. Still I can't understand the need for such a response (to 40kHz) while during recording the bandwidth is limited.

Chris

For those who haven't looked at it, I think that Stefanl's link is an excellent and very in-point medical/scientific paper. It documents a study in which "non-stationary" ultra-sonic sounds increased pleasurable brain activity in the test subjects, but only when sounds within the normal audio spectrum were simultaneously present. "We conclude, therefore, that inaudible high-frequency sounds with a nonstationary structure may cause non-negligible effects on the human brain when coexisting with audible low-frequency sounds."

So it would seem like ultra-sonic frequencies can be "audible," but only as a result of some intermodulation process with lower frequencies, or else, as the paper put it, "participation of nonauditory sensory systems such as somatosensory perception also needs to be considered in further investigations."

Still I can't understand the need for such a response (to 40kHz) while during recording the bandwidth is limited.

I'm not particularly familiar with the roll-off characteristics of professional recording microphones, but I would expect that although their bandwidth may only be specified to 20kHz in many cases, the roll-off would be gentle enough to capture significant content well above that frequency, assuming it is present.

So, if you are want to transport a 16-b word of music @ a 44.1KHz rate - this is what a CD laser mechanism does: every 1/44.1KHz seconds it spits out a 16-b word read off the physical spinning CD - you would have to transport each bit in 1/(16-b * 44.1KHz = 705.6KHz) seconds (so that you are ready to transport the next 16-b word that will arrive 1/44.1KHz seconds later.
So, your USB cable needs to have 705.6KHz bandwidth (& your DAC needs to run at 2*705.6KHz, as per Nyquist's criteria).

Keep in mind that two channels are present. That is the reason for the factor of 2 (which actually will be a little greater than 2 because additional non-data bits are needed to support the communication protocol).

Also, I'm not sure it's clear to everyone that the Nyquist criteria (sampled data systems being able to handle frequencies no higher than 1/2 the sample rate) does not relate to filtering at the output of a dac. Ultra-sonic spectral components that are present at the output of the dac chip itself, due to the 44.1kHz sample rate, will not alias, or "fold-down" to lower frequencies. A brick-wall anti-aliasing filter, which as I mentioned is needed at the input to an a/d, is not needed at the output of a d/a, even a non-oversampling d/a running at 44.1kHz. The spectral components associated with the sampling will be at frequencies of 44.1kHz and its harmonics (88.2, etc.), or at much higher frequencies if oversampling is used. They needn't necessarily even be filtered at all (consistent with Paulfolbrecht's comment), or if they are filtered the roll-off can be gentle.

Re Paul's comment about the desirability of no filtering, btw, I would expect that to be dependent on the specific components (preamp, power amp, speakers) that are being driven by the signal (particularly their bandwidths and their sensitivities to intermodulation distortion at high frequencies).

Regards,
-- Al

Bandwidth is not limited,which John Atkinson illustrates here, http://stereophile.com/features/282/ He closely looks at the very issue of what is captured on vinyl.Doug Rife postulates a theory of why this is important, http://www.mlssa.com/pdf/Upsampling-theory-rev-2.pdf both of these articles are good reading if you are interested and James Boyk's work at the Caltech Institute http://www.cco.caltech.edu/~boyk/spectra/spectra.htm looked at what ultrasonic content is captured by microphones until the signal starts to break-up at around 50Khz.