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Understanding Sample Rate


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2 minutes ago, tmtomh said:

 

Calling hogwash on digital sampling theory is like calling hogwash on the theory that a feather and a bowling ball dropped from the same height will (in the absence of air resistance) hit the ground at the same time. It's not a matter of opinion; it's a proven scientific theory. If digital sampling theory were untrue in the way you are suggesting, then digital music systems simply wouldn't work.

 

 

 

DSD samples higher than CD, right?

So everyone in this thread believes DSD is hogwash, right?

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2 minutes ago, bobbmd said:

@beerandmusic I have skimmed through this thread you started and you seem to have answered a questions(?s) I have have asked in various other threads and on different forums particularly ROON Community. I suddenly only have 16/48kHz through my delta sigma(pre multi bit) Gungnir yet I can tell or hear no real significance in SQ then when it gave me 24/192 or 24/176.4 and I commented is all this concern about sample rates BS since our ears basically can't hear better than 16/44.1. Yet at the same time I 'thought' the sound improved vastly when I got my first Modi and then even more when I got the Gungnir-at the same time when I went back and listened to my SACD and DVD-A recordings(the actual discs not ones I ripped) THEY sound better than anything I get now 'streamed' to my ears. Now when comparing the same song/album using my ME2(giving me MQA or non MQA) at anywhere from 24/48 24/96 etc and up or my DFRed at either 44/88/or 96 regardless of the 'player' used TIDAL or Qobuz desktop app or ROON or A+3 they all sound the same more or less( are my ears/brain 'burnt in' as someone else on this thread stated?).

Bottom line is I agree with you. Have we all been duped or has our perception of SQ been manipulated by God forbid FAKE NEWS or the Russians and everyone else on this august forum . It ALL sounds pristine to my ears and I may not buy anything new or better ie a multibit Modi/Gungnir(my CFO will be very happy).

bobbmd

 

I believe Man doesn't give God enough  credit (wink).

I think men are simpletons with big egos.

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8 minutes ago, crenca said:

 

And sampling at 44.1khz fully captures this waveform...fully...

 

it can't capture it accurately because it doesn't even take samples when much of the changes are taking place....and the only rebuttal can be the time it takes to transition...but the transitions are taking place infinitely as well.

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12 minutes ago, crenca said:

 

Sample rate does not describe the points (i.e the discernable differences), but rather the rate of change in the waveform - and it calculates this fully (i.e. there is no error) and thus it fully describes the waveform.

 

Think of it this way - you are counting the atoms - no, the subatomic particles in the paint of the Mona Lisa - you have forgotten what a painting is - she is a waveform, not the particles themselves...

I didn't ignore this...i will come back to it later....

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1 minute ago, crenca said:

 

It does not have to - it only has to describe the rate of change.  Your assuming that because a change takes place, that it is an unordered change, a random change.  If that were the case, it would not be a frequency, a waveform, a sound.  Your describing heat, not sound (and yet, even heat can be measured).

there is not enough bits in the world to describe the rate of change in a complex waveform with an infinite number of frequencies, even in the single pluck of a guitar string, let alone 10 million singers.

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4 minutes ago, crenca said:

 

Complex waveforms, as you imagine them, don't exist.  Remember how this conversation started, folks were correcting you on what a waveform (a "frequency") actually is.  It is simple (one), not complex (many).

 

You never really answered my question - how many waveforms do you hear when 3, or 300, or 3,000,000,000,000,000,000,000,000,000 singers sing at the same time?

 

one composite waveform with a bazillion different frequencies changing in an unordered fashion in a gazillion different times

 

if you are defining the waveform as one waveform per song, but i will leave that up to you.

 

 

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ok, correct me

3 minutes ago, crenca said:

 

If he granted the existence of waveforms, it would be.  However, he does not so what he is really describing is an infinite number of points moving randomly...is that quantum noise?

 

ok, correct me, on this simple exercise in correct verbiage.

 

a song has a guitar and a vocalist.

the singer sings 10,000 different frequencies in one song and plays the guitar with 20,000 different frequencies.

 

one song = one complex waveform with a multitude of different frequencies always changing and in an unorganized manner (e.g. singer sneezes in middle of song).

 

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5 minutes ago, crenca said:

 

All those "frequencies" (here I am misusing it as you are ;) - like it is a description of a point or a particle ) are related how?  Through time.  All these "frequencies" occur at the same point in time?  Nope.  How many frequencies do you hear at the same point in time?  One or many?  Is even a complex waveform (such as sound - it is three dimensional) continuous?  Yes.  Is the rate of change continuous or random?  Continous.  Can rate of change be described?  Have you ever taken calculous?  Can the area under a curve be measured fully?  If not, how does that bridge you drive over every day not collapse under the pressure of random, point like, infinite change?

 

You have a particle like, infinite mental image of waveforms and sound, which is simply not reality.

 

You hear a continually changing composite, with the gaps between samples averaged based on probability of samples, to keep the waveform continuous, but not based on factual samples between times?

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1 minute ago, tmtomh said:

 

You started out with a genuine curiosity in this thread, but now you're getting defensive and seem to be lashing out wildly. DSD samples much higher - but as @crenca notes above, it samples at only 1 bit of bit-depth!  DSD's native dynamic range is only about 6dB - that's six, not 60. Aggressive noise-shaping shifts most of all that horrible noise up beyond the range of human hearing, to around 28kHz or 30kHz if memory serves. But the point is that DSD's higher sample rate does not allow it to capture frequencies in the human hearing range any better or more accurately than PCM sample rates.

 

I have only considered frequencies within normal hearing ranges...i believe higher resolution provides more accuracy within the hearing range.  (e.g. i am, and only have been, concerned with what is within the audible frequency range).

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1 minute ago, crenca said:

 

A waveform has no "gaps", it is a rate of change - not a point like "here, then here, then here".  A waveform is not a stepping on individual stones (say in a river you are crossing), it is like the water flowing in the river.  Your confusing sample rate with rate of change.  Because human hearing (and sound itself through all mediums) is band limited, its rate of change can only be so much.  Thus, you only have to sample so much - you are thinking that the sampling is sampling the sound itself which you imagine to be particle like movement - it is not, it is sampling a waveform that just happens to exist in the particle based medium, which is like the flowing river and thus you do not need to sample every molecule of river water to measure the rate of flow.

 

what i meant by gaps is that at t1 you have freq x and t2 you have freq y, and you must connect the dots, so the detail between the dots is the gaps that is estimated, calculated, averaged, or whatever terminology you use....and that is where the details and accuracy are lost....between the samples.

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1 hour ago, crenca said:

 

Nope, because a waveform is calculated, not "sampled".  It is continuous...

 

in my thinking there is the real audio, the recorded audio and the playback audio all of which are different and only the real audio is accurate ....

 

either way, i will see if i can dedicate more time to it and see if i can get a better understanding, but my logic tells me that only what is real is accurate, everything that is constructed by man using his tools is very marginal,, and can always be improved on....i believe we have more information today than we had yesterday, and that music doesn't end at 44.1khz.

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1 hour ago, tmtomh said:

 

Your view is understandable - it's the most basic and widely shared misconception about digital sampling. But as @crenca just noted, analogue waveforms are reconstructed via calculation. There are no gaps because nothing unexpected, no "extra detail" is happening "in the gaps."

 

Here's an example: Say you're sampling at the CD rate of 44.1kHz. You've got a sound that's 15kHz. A 44.1k sample-rate system can encode that sound digitally (by taking at least two samples of it), so that the digital to analogue converter on the other end knows exactly what frequency that is supposed to be.

 

Now, there are "gaps" in the sampling - the digital sampling system does not sample the continuous, analogue waveform. Instead, it samples it "only" 44,100 times a second. 

 

But - and this is the key - by definition nothing weird, unexpected or "extra detailed" is happening in those gaps: it's just sound waves. The only way something could happen within the gaps that the sampling system wouldn't "know" about, is if you doubled (or quadrupled, or whatever) the frequency. In a 44.1kHz sample-rate system, 15kHz and 30kHz are encoded identically. The 44.1kHz system does not "know" that the alleged 15k wave actually is a 30k wave that oscillated twice instead of once during the "sample rate gap time." Similarly, a 60k signal also would look the same as a 30k and 15k signal to a 44.1k sample-rate system: 44.1k is not fast enough to sample each of those waves enough times to tell the difference between them. That's why the signal has to be bandwidth (frequency range) limited to a max frequency of 1/2 the sample rate. If it's not filtered, then in the above example the 30k signal would be "aliased," decoded by the DAC as a 15k signal, adding sound in the audible range that was not present in the original signal, aka distortion.

 

That is why I and others keep trying to tell you that sample rate correlates not to "extra detail" but rather to higher frequencies: the extra detail" can only come in the form of higher frequencies.

 

Finally, here are a bunch of links - they're pretty much random, the first half dozen or so that came up when I googled "is a higher sample rate better." They all say the same thing: Higher sample rates allow the capturing of higher frequencies, and they give more "cushion" for the necessary bandwidth limiting filters. Not a single one says that higher sample rates capture the 20-20k range more accurately or in more detail - and two or three of them explicitly state that this is not the case..

https://sonicscoop.com/2016/02/19/the-science-of-sample-rates-when-higher-is-better-and-when-it-isnt/

https://www.soundonsound.com/sound-advice/q-should-i-use-high-sample-rates

http://productionadvice.co.uk/high-sample-rates-make-your-music-sound-worse/

http://carriagehousemusic.com/analog-mixing/does-a-higher-sample-rate-audio-really-mean-better-quality/

https://www.sweetwater.com/insync/7-things-about-sample-rate/

thanks for sharing this...i will read tomorrow....overlooking it quickly i see it does touch on some areas i need to understand more.

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Ok, so i searched youtube for sample rates and found this video:

 

 

It states during the presentation, that not only does a higher sampling rate allow you to capture higher frequency rates, but it also allows you to take more samples to represent our audio (see time 2:50 seconds into the video).  
This is the part that I am concerned with...i don't care about higher frequency rates as a concern.

 

any way just now investigating if the logic I am trying to express is documented.

 

Again, i am NOT concerned with sampling rate to allow higher frequencies....

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So I just woke up, and haven't researched any more, and i see others have written more for me to read, but as of right now, this is my understanding.

 

Increasing the sample rate does 2 things.  ONE is related to norquist theorem and talks about the frequency range, which I have always stated that I don't have a problem with, because I am ONLY talking about what is in the audible range of hearing, and for the purpose of this argument, i am perfectly fine to just talk about frequencies between 600 and 700 hz.

 

The second thing that increasing the sample rate does, is along the horizotal axis and the accuracy of the resultant waveform by "connecting the dots".  This is what SONY was referring to about accuracy which i believe has nothing to do with the Norquist theorem.

 

So i have only studied for about 30 minutes, and about where i was last time i researched this and quit.

People can suggest I am thick, but I know otherwise, so it really doesn't bother me.  I know my IQ and ability to learn.

 

Logic also tells me if there was no possible truth, so many engineers would not even bother with sample rate higher than 44.1khz.  Clearly it is not a money making opportunity for everyone.

 

What i don't understand is why we can't stop talking about norquist and it's association with the highest frequency range, and lets just talk about the actual sampling on the horizontal axis and how more samples allows for more accuracy in the "connecting of the dots".

 

I will study more, but right now, i actually believe MORE about what i already previously believed.

 

If this is tiresome for anyone, i suggest you just bow out and ignore me and consider me ignorant...trust me, you won't hurt my feelings.  I would much prefer that, than to have to resort to name calling.

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7 minutes ago, STC said:

 

SACD is better than CD. The only problem I can’t reap the benefit.

 

All I need is speakers that are capable of producing the dynamic range of SACD. So I am looking at speakers capable of at least 160 dB 3 meters away taking into consideration of noise floor of 40dB.  

 

I also need an amplifier that could deliver about 3.5 million watt per channel to drive a typical 94dB sensitivity speakers. Any amps you want to recommend?

 

Clearly a mocking response, so i will give you a mocking answer....

 

You can't go wrong with McIntosh.

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23 minutes ago, Blackmorec said:

 

From which digital signal do you think your DAC would produce the most accurate analog rendition?

 

FIG.1: If the bit depth is low (a), the signal will be inaccurately converted because it’s sampled in large increments. By increasing the bit depth (b), you get finer increments and a more accurate representation of the signal.

 

The reason for using higher sampling rates isn't to produce higher frequencies, its to produce a more accurate analog output that better matches the original analog input. In simple terms, your DAC is doing less guesswork

 

From what i read yesterday, it does both....most of these people are tied to just part of what increasing the sample rate does, while totally disregarding the other thing it does....moving forward, i will only discuss what increasing the sample rate does across the horizontal axis.

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8 minutes ago, esldude said:

\ Watch the Digital Show and Tell video. You'll start to get somewhere if you'll understand it.

 

 

 

provide the link again....does it also talk about the improved accuracy by the higher sample rate?  That is where my focus will be.  My focus will not be on the highest possible frequency range, but on the accuracy provided by the higher sample rate.

 

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3 minutes ago, jabbr said:

Look you are trying to be understand this by “thinking about it” without understanding the underlying math. To me this all sounds like you are trying and trying to argue that 1+1=3 ??‍♂️

 

I do NOT want to talk about the highest possible frequency range, i want to talk about the accuracy of transitions.

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Just now, jabbr said:

Increased accuracy by higher sampling rate is not present in PCM encoding the way you are imagining. Until you have a solid understanding of the basics, topics like multibit SDM are going to be hopeless.

 

ok, well i am willing to put a couple hours into it...and see if my understanding is any better....and if not, people that suggeset SACD is not superior to CD, can just consider me ignorant even though my ears and my logic tell me differently.

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6 minutes ago, esldude said:

why higher sample rates only provide for higher frequencies and not improved accuracy. 

 

Sounds a bit biased and not sure I would even want to watch it unless it is approved by an authority I trust....if they don't discuss improved accuracy at all, it is not somehting i would be interested in....it sounds like it is more trying to prove a point that to explain the fundamentals....at least the way you describe it.

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4 minutes ago, mansr said:

You can trust Monty on this. Or you can follow the derivation of the maths all the way from basic arithmetic if you prefer. This is generally called "getting an engineering degree," and typically takes a few years of full-time study.

 

Yea, i am definitely not interested in that (grin).  I only want to know why an SACD sounds better than a CD and why so many people are reluctant to accept that....i don't think i am willing to spend more than an hour or two, so I will likely be considered "remained lost" by those that don't believe an SACD can sound better than a CD.

 

I was a  highschool dropout and retired at 55.  I have always marched to my own drum....maybe I am just a fool...that's ok.

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