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Author Topic: 48 kHz vs 44.1 kHz sample rate  (Read 4721 times)

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Offline DSatz

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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #45 on: October 05, 2020, 08:10:38 PM »
No, if you're downsampling then you MUST filter first (or simultaneously), and there's no special, more straightforward approach that applies to simple integer ratios. You might just as well go from 96 kHz to 44.1 kHz (or any other, arbitrary rate, even if it's not an integer!) as to 48 kHz. There is no gain or loss in efficiency, accuracy, dynamic range, distortion, ethical purity, sexual desirability, followers on social media, or anything else either way.

If you're upsampling, then part of the code could be made simpler for the special case of integer ratios of sampling rates. However, no actual sampling rate conversion hardware OR software that I've ever seen, smelled, tasted, or heard of uses such an approach. I'm trying and failing to find a suitable metaphor to explain why this is, since metaphors seem to convince people where actual descriptions of reality do not.

Yes, those special cases would be easier to program than the general case. But the writers of the software or firmware would STILL need to support the general case as well, test it, AND THEN have the "easier" way standing by for whatever percentage of cases in which it could be used. They'd also have to build in additional logic to detect the special cases that could use the simpler code and switch over to it. But that's a worse situation for the software or firmware developer, not a better one. Yes? The inconvenience and risk that they would have to take on in order to code, test and integrate their implementation of the "easier" approach, would come in addition to the inconvenience and risk of handling the more general case that they have to handle anyway.

So no one that I've ever heard of, or can realistically imagine, does that when writing sample-rate conversion software.
« Last Edit: October 07, 2020, 03:48:30 PM by DSatz »
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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #46 on: October 05, 2020, 11:36:32 PM »
Yes, those special cases would be easier to program than the general case. But the writers of the software or firmware would STILL need to support the general case as well, test it, AND THEN have the "easier" way standing by for whatever percentage of cases in which it could be used.
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Offline Gutbucket

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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #47 on: October 06, 2020, 06:04:12 PM »
"I'm trying and failing to find a suitable metaphor to explain why this is, since metaphors seem to convince people where actual descriptions of reality do not."
^
Thanks for this gem dejour, which very much rings true.. but then, isn't all language inescapably metaphoric? 
Even mathematics, which seems odd. 
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Offline DSatz

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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #48 on: October 07, 2020, 02:47:15 PM »
I don't know about all language being metaphorical; maybe. This gets into Kant and Hume and whether or not we ever experience "reality itself". All I know so far is that I respect those questions, and can't answer them for sure.

And I understand the impatience of Ayn Rand and her followers, who refuse to respect those questions--which, however, doesn't make the questions go away; it just makes "Objectivism" another branch of religion, in which people choose what they will believe because of how it makes them feel when they believe it. Hulk smash: I want life and the universe to work a certain way, therefore I claim that they should and do work that way. First the verdict, then the trial ...

Yeah, so, metaphors and visualizations--they're a huge help, but they can also cement mistaken impressions into place (see that metaphor there?). Sooooooo many people imagine the sampling process in A/D conversion in a way that makes it "perfectly obvious" that the higher the sampling rate, the more accurate the reconstruction will be. And that's just not a fact, given the Shannon/Nyquist bandwidth constraints. But as long as you're thinking by reference to the visual metaphor, you can't see what's true and what's not true about that model or outside of that model, where the sampling and reconstruction processes actually live.

It gets to where people won't even listen to evidence from the real world, because their mental model is so precious to them. Not just occasionally, but really often.

I think that's a lot of why Aristotle was the official philosopher of Western Christianity for so many centuries despite being a pagan. His whole system was made up of assertions that feel right, as long as you don't have any empirical knowledge of the subject at hand. And back then, not many people did. As a result, empirical knowledge itself wasn't respected; think of the people who refused "on principle" to look through Galileo's telescope.
« Last Edit: October 07, 2020, 03:06:00 PM by DSatz »
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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #49 on: October 07, 2020, 04:38:54 PM »
I think that's a lot of why Aristotle was the official philosopher of Western Christianity for so many centuries despite being a pagan. His whole system was made up of assertions that feel right, as long as you don't have any empirical knowledge of the subject at hand. And back then, not many people did. As a result, empirical knowledge itself wasn't respected; think of the people who refused "on principle" to look through Galileo's telescope.

Of Medieval Judaism and (especially) Islam as well....
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Offline kuba e

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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #50 on: October 08, 2020, 06:03:01 AM »
My little vote is for Gutbucket. They taught me that language, words are concepts. For me, math and thoughts belong there too. Unfortunately, we need words. Ha ha, for most of my friends, "Shannon/Nyquist bandwidth constraints" is a boring word at best. But for me it's an interesting word thanks to Taperssection.

I would have a question if I imagine the following just like the others:

According to Shannon / Nyquist - 44.1kHz sampling is sufficient for accurate signal reconstruction when the recorded signal does not contain frequencies higher than 20kHz. My microphones are cheap and I can't hear even 15kHz. So 44.1kHz is perfectly sufficient for me.

For the reason of higher sampling, Wforwombo mentioned post-processing. Does this mean that some post-processing also takes into account frequencies above 20kHz and the processing of these high frequencies will affect sound that I already hear - 15kHz and below? I use only basic plugins like eq, limiter, compressor, sometimes multi-band compressor.

« Last Edit: October 08, 2020, 07:46:01 AM by kuba e »

Offline wforwumbo

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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #51 on: October 08, 2020, 12:02:41 PM »
On the whole, what DSatz has said since I last checked this thread is right.

I’ve seen resampling code that DOES have the “special integer case” resampling method in the field, BUT it is a safe assumption that any publically available resampling code is NOT doing this.

For the reason of higher sampling, Wforwombo mentioned post-processing. Does this mean that some post-processing also takes into account frequencies above 20kHz and the processing of these high frequencies will affect sound that I already hear - 15kHz and below? I use only basic plugins like eq, limiter, compressor, sometimes multi-band compressor.

This is a good question - but it’s confusing the symptom with the cause.

The “band limiting” of 20 kHz has to do with *accurately reproducing the signal in the analog domain*

Digital signals DO have data outside of that bandwidth of 0-20 kHz. They actually have a lot of information outside of that range defined from -infinty to +infinity, and even more technically the baseband has info from -20k to +20k (don’t think too hard about that one...) but this is another conversation for another day.

Here’s what Nyquist-Shannon are ACTUALLY saying in their theorem, paraphrased by me (I encourage you to seek out their 1927 paper and read it yourself, I’m happy to answer any questions and discuss differences in interpretation):

If you have an analog signal, and want to sample it into a digital domain, and then reproduce it back in the analog domain... in order to do so without any distortion or loss of information, you must sample the signal at a rate that is at least twice the highest frequency that is contained within the signal.

So you need to disentangle the “a 44.1k signal is only capturing up to 20k” from the sample rate. The sample rate is a property of your recording that has effects and impacts - one of them being, in a theoretically ideal world you could only reproduce data with a bandwidth of 0-22,050 Hz without distortion if you do everything else right.

The post processing argument has to do with a reduction in error, especially for filtering. The degree of audibility of that error is up for debate. But the idea here is that you’re no longer reproducing the original signal, you’re now modifying it - and that has impacts elsewhere in the chain. These impacts are a function - symptom of - sample rate, not necessarily a cause.
« Last Edit: October 08, 2020, 04:32:32 PM by wforwumbo »
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Offline kuba e

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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #52 on: October 08, 2020, 04:10:18 PM »
Thank you Wforwumbo. I think I am starting to understand. 44.1k sampling is sufficient for accurate reproducing a signal to analog domain, e.g. for playback up to 20k. But digital post-processing is modifying a signal in digital domain. And apply here, the more accurate the samples the better.

I have two more theoretical question:

Could we record at 44.1k/16bits and resample the signal later in a case someone would like to do a fine digital post-processing? If I understand correctly: The result should be very similar to the recording that is recorded from the beginning at 96k/24bits. The only difference is that there are two additional d/a and a/d converters in the recording chain. Is it right?

I know that DAW can internally resample signals into the higher rate because more accurate calculations. Is there a difference between an internal resampling in DAW and physical d/a and a/d converters?

I understood not all the posts in this thread, I apologize if anyone has already explained this.

Digital signals DO have data outside of that bandwidth of 0-20 kHz. They actually have a lot of information outside of that range defined from -infinty to +infinity, and even more technically the baseband has info from -20k to +20k (don’t think too hard about that one...) but this is another conversation for another day.
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« Last Edit: October 08, 2020, 04:37:15 PM by kuba e »

Offline wforwumbo

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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #53 on: October 08, 2020, 04:40:40 PM »
Thank you Wforwumbo. I think I am starting to understand. 44.1k sampling is sufficient for accurate reproducing a signal to analog domain, e.g. for playback up to 20k. But digital post-processing is modifying a signal in digital domain. And apply here, the more accurate the samples the better.

I have two more theoretical question:

Could we record at 44.1k/16bits and resample the signal later in a case someone would like to do a fine digital post-processing? If I understand correctly: The result should be very similar to the recording that is recorded from the beginning at 96k/24bits. The only difference is that there are two additional d/a and a/d converters in the recording chain. Is it right?

I know that DAW can internally resample signals into the higher rate because more accurate calculations. Is there a difference between an internal resampling in DAW and physical d/a and a/d converters?

I understood not all the posts in this thread, I apologize if anyone has already explained this.

Assuming ideal conversion between the analog and digital domain, I don’t think you’d hear a difference re: sample rate. Taking 44.1 content, converting it to analog, and then sampling that at 96 - assuming no distortion - should give the same result as native content at 96k. Then you could do your processing at the higher sample rate without issue. This is because the audible frequency range is captured at both sample rates with no distortion.

Bit depth is different; once the bits are gone, they’re gone. You can’t really replace them. IF the dynamic range of the original content is less than 96 dB AND the noise floor of your system is greater than 96 dB and induces no distortion, then I don’t think you would see an appreciable difference post-reconversion, but those are two big ifs.

The last question depends on the DAW and resampling algorithms of choice.

How much any of this is appreciable or noticeable is still up for debate. One of these days I’ll do an A-B...

Addendum: technically in the resampling process, there will be distortion between the two cases. But that distortion will occur between 22.050 kHz and 48 kHz. And if I were a gambling man I’d put money down that you couldn’t perceive such distortion.
« Last Edit: October 08, 2020, 04:51:27 PM by wforwumbo »
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Offline aaronji

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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #54 on: October 08, 2020, 04:55:13 PM »
Digital signals DO have data outside of that bandwidth of 0-20 kHz. They actually have a lot of information outside of that range defined from -infinty to +infinity, and even more technically the baseband has info from -20k to +20k (don’t think too hard about that one...) but this is another conversation for another day.

Would you be so kind as to explain this in greater detail? If not here then in a new thread? I can understand why this would be the case in the mathematical realm, but I can't wrap my head around the idea of negative Hz in the physical realm. It seems, basically, impossible to me.

Bit depth is different; once the bits are gone, they’re gone. You can’t really replace them. IF the dynamic range of the original content is less than 96 dB AND the noise floor of your system is greater than 96 dB and induces no distortion, then I don’t think you would see an appreciable difference post-reconversion, but those are two big ifs.

That first one is definitely not a big if for tapers. The maximum SPL of the music minus the ambient SPL is probably almost always less than 96 dB in our world. DSatz mentioned above that he (thinks he) has only encountered a recording situation that actually exceeded 96 dB once.
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Offline kuba e

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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #55 on: October 08, 2020, 05:23:38 PM »
Thank you very much Wforwumbo for nice explanation. I understand.

My question about resampling in DAW was meant with regard to - If someone has a good DAW with good resampling algorithm, can this replace d/a and a/d physical converters to resample the recording from original 44.1k to 96k due to fine digital post-processing?

Offline wforwumbo

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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #56 on: October 08, 2020, 05:33:13 PM »
Digital signals DO have data outside of that bandwidth of 0-20 kHz. They actually have a lot of information outside of that range defined from -infinty to +infinity, and even more technically the baseband has info from -20k to +20k (don’t think too hard about that one...) but this is another conversation for another day.

Would you be so kind as to explain this in greater detail? If not here then in a new thread? I can understand why this would be the case in the mathematical realm, but I can't wrap my head around the idea of negative Hz in the physical realm. It seems, basically, impossible to me.

Bit depth is different; once the bits are gone, they’re gone. You can’t really replace them. IF the dynamic range of the original content is less than 96 dB AND the noise floor of your system is greater than 96 dB and induces no distortion, then I don’t think you would see an appreciable difference post-reconversion, but those are two big ifs.

That first one is definitely not a big if for tapers. The maximum SPL of the music minus the ambient SPL is probably almost always less than 96 dB in our world. DSatz mentioned above that he (thinks he) has only encountered a recording situation that actually exceeded 96 dB once.

Re: the second point, for the vast majority of situations we are in, yes you are correct. I'll note that I've been in rooms with NC-5 (meaning that the "background hum" is less than 5 dB) and peak volume of a performance was at 127 dB-C, but that's definitely an exception to a rule. It was definitely interesting having the self noise of my microphones and the dynamic range of my mic pres be the bottleneck in the dynamic range chain... but again, exception to the rule.

Re: the first question... yes. More to follow later this evening when I have some time to draw out diagrams, since it actually gets at the core of why the sampling theorem exists.
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Offline wforwumbo

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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #57 on: October 08, 2020, 05:35:55 PM »
Thank you very much Wforwumbo for nice explanation. I understand.

My question about resampling in DAW was meant with regard to - If someone has a good DAW with good resampling algorithm, can this replace d/a and a/d physical converters to resample the recording from original 44.1k to 96k due to fine digital post-processing?

Short answer is "no", the less succinct answer is "with lots of caveats, it's possible", and the best answer is "just record at 96k in the first place to avoid such a headache."

There are tricks we can play in terms of "modeling the transfer from digital back to the analog domain" with upsampling, but our tricks can only get us so far. There are some standards that try to model this (such code is used to accurately display dB meters in DAWs, for example), but because of the shortcomings of resampling algorithms they're already inherently a bit flawed.

Edit: If your question is more about understanding the philosophy behind sample rates and resampling, then yes your intuition is barking up the right tree here. But in practice... just record at the higher sample rate.
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Offline aaronji

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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #58 on: October 08, 2020, 05:56:18 PM »
Re: the second point, for the vast majority of situations we are in, yes you are correct. I'll note that I've been in rooms with NC-5 (meaning that the "background hum" is less than 5 dB) and peak volume of a performance was at 127 dB-C, but that's definitely an exception to a rule. It was definitely interesting having the self noise of my microphones and the dynamic range of my mic pres be the bottleneck in the dynamic range chain... but again, exception to the rule.

I think there is a vanishingly small probability of that happening in a taper situation! Even in a studio setting, I suppose 5 dbSPL is a real rarity. For us, I think we are lucky if the background is only 50 dB higher...

Re: the first question... yes. More to follow later this evening when I have some time to draw out diagrams, since it actually gets at the core of why the sampling theorem exists.

I would definitely appreciate it! I think I have a pretty good layman's understanding of Nyquist-Shannon, but I like numbers and would appreciate the deeper insight.
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Offline kuba e

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Re: 48 kHz vs 44.1 kHz sample rate
« Reply #59 on: October 08, 2020, 06:54:22 PM »
Thank you!

I was asking because of both philosophical and also practical. I am doing amateur, audience recordings at 44.1k/24bit. I am archiving the original recordings after level adjustment at 44.1k/16bit. My dynamic range is lower than 96 dB. I don't record extremely loud music and the background noise at concerts is relatively high. I hope 16bit won't hurt. Probably 44.1k is also enough for me because I'm not good at post-processing, I am not able to distinguish fine details. But I will try to compare post-procesing with 44.1k vs 96k.

What's good I learned here is that if I find some treasure in the archive, whoever would do quality digital post-processing in the future can resample my original with good a/d convertors. And the loss will not be so big.

Of course, I understand now that those who do well post-processing or those who record professionally record and store at 96k/24bit.
« Last Edit: October 09, 2020, 08:36:40 AM by kuba e »

 

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