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How many bits, how fast, just how much resolution is enough?


BlueSkyy

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Word (as they say). ;)

 

 

As a lawyer, what I nearly always find when non-lawyers talk about the law is that they tend to leap to sweeping over-generalized conclusions. I'm sure something very similar occurs to you as you read what non-professionals write about filters, modulators, audio software, etc.

 

A failure of fundamental responsibility to serve the law, which serves the people. How we've gotten to where we are is a huge portion of the problem..........that it takes professionals to explain the difference between right and wrong.

 

You may agree.......or not.......inconsequential i suppose........but just a relevant as professionals suggesting we like A vs B. Debacle?.......I dunno?

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Of course it is. Imagine two equivalent 10 KHz pure analog sinusoidal tones, one started 100 microseconds before the other. A digital (PCM) 44.1/48 KHz sampled recording of this will sound different from a higher resolution digital recording..

 

It's important to look at these concerns in relative terms and conditions. In the case of your example, given the wavelength of 1.35" at 10khz, if your head at the time of listening or your speakers in the horizontal plane are off axis 1 degree or more, you've already exceeded your 100 microsecond time smear by a magnitude of 3 or greater.......and this assumes laboratory consistency for everything else in the signal and environmental chain.

 

I agree, you can make an argument for anything, but relevant arguements are few and far between. If i take 2 samples of a high end tweeter, install them on the same fixed test baffle without changing anything in the signal chain or testing environment the two responses, CSD, distortion sweeps, phase measure or impedance will NEVER measure exactly the same. Add humanistic variables like blood pressure and glucose levels and the waters get even murkier for your position. Coefficients of significance.

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Typical time domain spread of a RedBook oversampling filter is around 1 millisecond.

 

Only working with his example buddy......but even for a millisecond (.001) there's still no relevance unless ones head is clamped in a vise of chair positioned in a measured equilateral triangle in an anechoic room of equandestant dimensions.

 

More of a matter of preference for me but I tend not to listen under those conditions! Lol

I like to move around a bit, ya know?

 

God help us all if the culinary world develops the same audiophobia criteria where everything matters! Who wants a lab technician instead of a Michelin chef?

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Likely yes... I have paid particular attention to use equipment that can cleanly reproduce at least to ~50 kHz.

 

Some of the improvements are audible also without such equipment, for example improvements in intermodulation and time domain behavior. Also in many cases jitter and level linearity performance improves.

 

Again.......I've figured after a few years of this some of you would have stopped trying to separate time from frequency in an effort to incorrectly attribute suspected audible differences to this mystical place. Can't have one without the other fellas! If timing changes, so does the response at the listening position.

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BTW, most 96 khz and above devices don't use the opportunity to use a less steep filter. They just extend the bandwidth and keep steep filters. Lavry does use a gentler filter starting to roll off above 30 khz when doing 96 khz sampling in his gear. Maybe filtering that starts at 30 khz on DXD would be great. I take it Lavry thinks you gain nothing.

 

Easier to track phase and tighten impulse response this way with the steep filters. Shallow slopes present their own problems of course.......like huge files for one

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This is basically where I am with Hi-Res. I am -very- happy with quality recordings whether they come from Tidal, a CD rip or a Hi-Res d/l. I currently run everything through Roon, up-converted to DxD (or DSD128) in HQ Player and sent to my µR-->DAC. I'm hesitant to pay more than the CD price for a Hi-res version unless that particular version is not available otherwise.

 

My Man!.......couldn't have said it better myself!

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In my view, the problem with using audibility as a gauge is that there are aspects of equipment performance that can be affected by out of audible range noise, such as the tweeter example that I mentioned previously and if I am not mistaken the fact that some amplifiers are negatively affected by high in level supersonic noise.

 

Besides, the cumulative result of a few tiny inaudible improvements may end up being audible.

 

R

 

This assumes the production of modal harmonics from those signals outside of our range and is really easy for you and others to test yourself.......just disconnect all the other drivers in your speakers except your tweeters with response to 50khz and play loud test tones of the frequencies in question and see if those theoretical harmonics are audible?........

 

This also assumes that the recording microphones, sources and mastering and mixdown gear were not electronically capped in range above 20khz or whatever FR you suppose proves your hypothesis. Be interesting to see what common era ADCs do in this regards?.......is that content filtered out in the analog stage, digital, both?

 

Homework assignments!.......yes!

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