[ProAudio] The High-Resolution Challenge

Sat Feb 15 16:15:21 EST 2020

Let me add something to the last email. The reason to up sample is to enable easier filtering. Almost all DA converters already up sample a lot. If you feed a DA 48Khz , and it needs to operate it say 256fs, feeding the same device 96Khz may require only 128 upsampling. You get to skip the first X2 stage. So if you hear a difference, it will be about implementation of that first stage. It is not about higher frequencies, it is about anti imaging filter.It does not reduce audible distortions...RegardsDan LavrySent from my Verizon, Samsung Galaxy smartphone

-------- Original message --------
From: Dan Lavry via ProAudio <proaudio at bach.pgm.com> 
Date: 2/15/20  3:47 PM  (GMT-08:00) 
To: Mike <mm1100 at yahoo.com>, proaudio at bach.pgm.com 
Subject: Re: [ProAudio] The High-Resolution Challenge 

When talking about hi definition standard, do we include our pets?Your comment about more samples yielding better results is incorrect. Regarding the upsampling comment,if your samples are perfect, you would not need more samples. If your samples have errors or noise, the new samples computed from the original set are going to contain the errors from relying on the original samples (with errors). You can't erase the errors, or the noise, there is no such magic.I do not argue with anyone about what they hear. But I don't think that anyone is going to claim that they can hear a loud 40Khz sine wave. Let's make it simple, forget digital audio for a moment. Take a sine wave at 40Khz, with the trancducers you mention. Do you think a human can hear it? I think it is reasonable to expect some agreement regarding what is acceptable audio bandwidth for the most capable ears. How can anyone avoid it in a conversation about hi definition audio?I am old. I saw how 1KB ram made room to 40GB. There was AM radio and we now deal with many GHz... But to the best of my understanding the ear bandwidth has not seen much change...RegardsDan LavrySent from my Verizon, Samsung Galaxy smartphone-------- Original message --------From: Mike via ProAudio <proaudio at bach.pgm.com> Date: 2/15/20  12:22 PM  (GMT-08:00) To: proaudio at bach.pgm.com Subject: Re: [ProAudio] The High-Resolution Challenge On 2/14/2020 5:28 PM, Dan Lavry wrote: > I am not saying that 48KHz is better or worse than 96KHz. Clearly at 44.1Khz > the theoretical bandwidth is 22.05Khz. That is already a couple of dB loss at > 20Khz. Add a 20Khz mic for another 3dB, and perhaps a speaker for another > 3dB.. pretty soon you find it to be difficult to accomodate the 20-20Khz hi fi > standard.At the time when this pipe-at-a-time organ company was in business, Sennheiser was touting a mic that was only a couple of dB down at 50 kHz. And if you're also supplying the speaker system for the organ, there are a number of transducers that get up into the ultrasonic range to take over where the cone loudspeakers poop out. I'm not arguing that this is all a great idea, but it's feasible. And when the ultrasonics mix in air, you get something in the audible range. > It seems to me that too much is attributed to figuring sample rate, which is a > step removed from the question of what we need for bandwidth. > I undestand that it us not intuitive to understand that you only need to reach > some minimum number of samples for a given bandwidth, and more samples > do not add any information for perfect reproduction of the original.Well, in this case, the folks building the organ had a rational case for needing extended bandwidth to do what they wanted to do. I record fiddles and banjos, and I don't miss anything at 44.1 kHz.One case for higher sample rate is that for any given frequency, the samples are closer together, so for the standard audio bandwidth, you have twice as many samples as Shannon and Nyquest say you need in order to perfectly reconstruct what went into the system. In a universe where we do a lot of digital number crunching to emulate filters and dynamics processors (ahem! non-linear elements) you get less rounding error when you average more points. This is an explanation (not sure if it's the correct explanation) of why up-sampling can make a standard sample rate converter sound better. > I wrote a whole 32 page paper Sampling Theory, based on Nyquist work, but > some folks can't shake the notion that the more is better....I read that years ago, probably still have the PDF somewhere. I remember thinking that you were absolutely right then, and you still are today. The point is that because we can sample faster than Nyquist says we need to, we can come up with situations where there's content above the standard audio band that, when taken out of isolation, does make a difference when sounds interact.[added to my reply to Dan after reading the next blast of digest]Apologies to Dick and JJ regarding the non-linearity requirement for the supersonics to combine and make a difference (both meanings). That's how they explained it to me and I didn't know enough to argue about it. I didn't realize that air was that linear at respectable SPL. But if they weren't convinced that it mattered, they could have saved a lot of money on hardware and disk space, as this was before the days of the $29.99 2-terabyte drive.-- For a good time call http://mikeriversaudio.wordpress.com_______________________________________________ProAudio mailing listProAudio at bach.pgm.comhttp://bach.pgm.com/mailman/listinfo/proaudio
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://bach.pgm.com/pipermail/proaudio/attachments/20200215/7a778074/attachment.html>