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In Reply to: RE: Food for thought posted by Jaundiced Ear on September 14, 2014 at 22:30:32
I'll go and read up some more on exactly what gets sampled and when. I'm basing this on the premise that everything gets sampled, and increasing sample rate also increase sample range.
I appreciate the analogue is a single continuous entity but the breaks do exist, in the digital at least. If the DAC is knitting the seams together, or filling the intervening space with algorithmically estimated data, the result is an analogue signal that could be improved, depending of course on what gets sampled when. I'm off to do some more reading.
Regards
Follow Ups:
Hey Storris!
Remember, we're dealing with bandwidth limited signals. That's one of the keys to digital: we're not dealing with infinite bandwidth, but only a limited set. The theory does sound crazy, but you don't even think about it when you hear voices coming out of your telephone or watch a TV show.
JE
The theory is that the signals are bandwidth limited. Theory is only an approximation to reality because a finite duration signal can only be approximately band limited. For a system to work well in practice there must be a significant guard band between the audio signals and the theoretical bandwidth determined by half the sampling rate.
Also, just to set things straight from another post in this thread, if one has a sine wave it is not sufficient to sample it twice per cycle. One must sample it more than twice. (To see this, consider that the two samples might just have hit the positive and negative zero crossings of the sine wave.) If there are only slightly more than two samples, then the filter has to be very sharp, e.g. "ring" for a very long time, to figure out what the original sine wave was.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Thanks for the links by the way, very informative stuff...I get what's being said, and think it sounds perfectly sane when applying it to a constant tone, but not with music. As far as I can tell, there is no physical way that 48,000 images of a thing can give you a picture as complete or accurate as 3,000,000 when that thing is changing at an infinitesimal level, constantly. Details will be missed.
This is true for video and still images in digital and analogue, even eyesight, so how not for A/D/A sound?
Anyway, I've put some questions up on his Wiki.
EDIT
Unless the Bit Depth is involved? 16Bits/MoreFrequencies = MoreFrequencies per Bit = Less detail than 16Bits/FewerFrequencies. But Bit Depth is Dynamic range...?
Also,
In response to other questions on the video's wiki page, he or someone at least, has written this;
"Does this mean that we should have better output if we increasing the sampling from 44.1KHz to 192KHz?"
"You'll have exactly the same 20 kHz sine wave at 192 kHz as 44.1 kHz... Only two sample points are needed to perfectly recreate a sine wave... --Leorex"
"It's counterintuitive, but try and think of it like this. You know the input signal (analog) is band-passed to 20kHz, so there are no frequencies higher than 20kHz to be reconstructed. Now look at the 2.2 samples per period; try and draw a continuous line through all the samples without using any frequencies above 20kHz. So in fact, there is only *1* solution for the line you draw through the sampling points. You can 100% recreate the original analog signal from the sampling points. You will not get better output by increasing the sampling rate to 192kHz, because you have already reconstructed 100% of the signal. -- Nhand42"
To my mind these answers only suffice for constant frequency/amplitude signals. When you add a constantly changing frequency/amplitude signal, the ability to "draw a continuous line through all the samples" will become much more difficult. We are not trying to recreate a Sine Wave. Try illustrating an orchestral piece with his lollipop diagram.
Edits: 09/15/14 09/15/14 09/15/14 09/15/14 09/15/14 09/15/14 09/15/14
Sorris:
"I get what's being said, and think it sounds perfectly sane"
No, if you got what was being said you would not think it sounds perfectly sane.
"I get what's being said, and think it sounds perfectly sane when applying it to a constant tone, but not with music. As far as I can tell, there is no physical way that 48,000 images of a thing can give you a picture as complete or accurate as 3,000,000 when that thing is changing at an infinitesimal level, constantly. Details will be missed."
You think this because you have no understanding of how digital audio works. Perhaps this will help.
Way back when, people wanted to record analog events around them (people speaking, or making music}. They only had analog tools so they used them. These analog tools created a one-dimensional analog wave form that people could fiddle with and send over the air, or impress onto vinyl, or use to magnetize iron paritcles glued onto tape. It didn't really matter. Those formats were all used to create analog waveforms that were then amplified by amplifiers and used to drive speakers.
The crucial point to remember is that a signal consisting of a one dimensional, varying voltage, was used to capture and to simulate the sound of music or other noises.
Digital Audio is NOT a description of reality. Digital Audio is a description of the one dimensional, varying voltage that, until now, has been used to simulate sound and music.
So long as you keep trying to use "Digital Audio" to describe "reality" you are going to go insane. Only once you get the idea that "Digital Audio" is only useful to describe "recorded audio" will you get to be comfortable here.
All the Best!
JE
"varying voltage"
Two words that have fixed everything, cheers.
You just needed a new perspective. Just as there are no "quantum speakers" neither are there any "quantum microphones." When you think about it, is there any part of recorded audio, digital or analog, that can't be attributed to a voltage variation?
JE.
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