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Understanding Sample Rate


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15 minutes ago, beerandmusic said:

no i didnt know that...but isn't that one of the reasons to increase the sample rate to move this filtering outside of the audio band?

 

Or if you want to expand on why it makes sense to increase the sample rate related to filtering, some may be interested?

 

I personally have already been content some time ago.

That is why the redbook sampling rate is 44kHz (22Khz Nyquist), so that the filter cutoff can be above the audible band.  Philips and Sony really did think about these things.

Edit: maybe this will help https://en.wikipedia.org/wiki/44,100_Hz

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7 minutes ago, psjug said:

That is why the redbook sampling rate is 44kHz, so that the filter cutoff can be above the audible band.  Philips and Sony really did think about these things.

 

What is it that people are talking about when they suggest upsampling to allow the filter procesing outside of the audioband to improve SQ?  Not that i really want to know...but that is what i was referring to....

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47 minutes ago, adamdea said:

Possibly as regards bandwidth- but just as easy to ignore if one takes the fluctuations of one's listening experience and one's personal interpretation of the meaning of those experiences as the only critical data, alongside a general faith in progress.

The way I see it, the goal of audio engineering should be to accurately reproduce the stimulus a listener is subjected to, i.e. the sound waves entering his or her ears. The study of perception belongs to the domain of neuroscience (or something similar). The only way it relates to the engineering is when its findings allow engineers to take shortcuts. A prime example would be the discovery of two-channel stereo and its ability to create the illusion of arbitrary left/right positioning without using a speaker for every possible direction.

 

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At one point it looked as though DXD might become a de facto "this far and no further" standard. But forget it. Limits are offensive and if you play someone a dxd file and a 32/ 768 file pretty much the same group of people who can hear the difference between 16/44 and 24/96 will claim to hear the improvement in the 32/768. 

Don't I know it.

 

Quote

When first came to this hobby I assumed that the 16/44 spec must have been cobbled together as a result of the limits of technology in the early 80s and that it made sense that it must be easy to improve on it now. After all I remembered what computers were like in 1982 and assumed that analogies with video were closely applicable. It took me a long time to "unlearn" that.

The process leading to the specific design of the CD is interesting. The rough requirements were determined by the already well known limits of human hearing, both in frequency and dynamic range. Sony and Philips argued a bit over whether to use 14 or 16 bits, mostly because one of them had a 14-bit DAC chip and the other had a 16-bit one. The choice of precisely 44.1 kHz as sample rate was made because existing equipment could easily be made to handle it. The physical design of the disc (laser wavelength, track spacing, etc.) was mostly based on what could be manufactured at a reasonable price. As for the size of the disc, one story has it that it was chosen such that the longest known recording of Beethoven's 9th symphony would fit.

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24 minutes ago, psjug said:

That is why the redbook sampling rate is 44kHz (22Khz Nyquist), so that the filter cutoff can be above the audible band.  Philips and Sony really did think about these things.

Edit: maybe this will help https://en.wikipedia.org/wiki/44,100_Hz

Yes, but at least on some recordings there could well be material above 44.1k, musical or otherwise. Recording at 96k or above eliminates this as a potential problem. 

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22 minutes ago, mansr said:

The way I see it, the goal of audio engineering should be to accurately reproduce the stimulus a listener is subjected to, i.e. the sound waves entering his or her ears. The study of perception belongs to the domain of neuroscience (or something similar). The only way it relates to the engineering is when its findings allow engineers to take shortcuts. A prime example would be the discovery of two-channel stereo and its ability to create the illusion of arbitrary left/right positioning without using a speaker for every possible direction.

I think there are a number of rational approached which could be taken to this. I think quite  a lot of engineers would say that the goal was to reproduce everything you can hear. Others might say that it is to maximise the verisimilitude of the recording

 

Either way I am surprised at the extent to which people get worked up at any shortcomings of the 16/44 channel compared with the shortcomings of only having two channels, the need to reproduce the lowest audible octave,  and the problems caused by rooms.

 

Taking your goal of  "accurately reproduce the stimulus a listener is subjected to, i.e. the sound waves entering his or her ears" I'm not 100% sure you can dispense with any psychoacoustics. I assume there are two ways of reproducing the sound waves in the ear: 

One characterised by ambisonics would be to measure the 3d vector and then try to produce that in the listening room; the other is to pre-treat the sound with the hrtf and pipe it to the ear

 

I'm not convinced that you could ever perfectly achieve the reproduction part of the ambisonic soundfield , so I suspect you would still need a theory of what really matters. I also suspect that perceptual science would still have something to say about the limitations even of a "perfect" reproduction though. Because you aren't in the recorded acoustic space  it's difficult to "learn" it.

Similarly with the hrtf it is psychoacoustics that tells us of the need to have head tracking isn't it? And in the absence of perfect hrtf measurements (possibly matching the clothes I am wearing today) you need a theory of what really matters.

 

You are not a sound quality measurement device

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23 minutes ago, mansr said:

The process leading to the specific design of the CD is interesting. The rough requirements were determined by the already well known limits of human hearing, both in frequency and dynamic range. Sony and Philips argued a bit over whether to use 14 or 16 bits, mostly because one of them had a 14-bit DAC chip and the other had a 16-bit one. The choice of precisely 44.1 kHz as sample rate was made because existing equipment could easily be made to handle it. The physical design of the disc (laser wavelength, track spacing, etc.) was mostly based on what could be manufactured at a reasonable price. As for the size of the disc, one story has it that it was chosen such that the longest known recording of Beethoven's 9th symphony would fit.

Yes very interesting and well covered here https://www.amazon.co.uk/Perfecting-Sound-Forever-Story-Recorded/dp/1847081401

You are not a sound quality measurement device

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3 hours ago, jabbr said:

Oohh ... a system that produces IM distortion with information only above 20 kHz is not so well designed. 

 

Products of intemodulations are generated by full band: audio signal + ultrasound.

 

When we cut ultrasound, we remove part of the intermodulations.

 

But intermodulation products that produced by audio band are remains.

 

Intermodulations are harmonics with frequerncies by formula:

 

Fintermod_product=N*f1+M*f2,

where

M, N are positive or negative integers,

f1 and f2 pair of harmonics, that contains into the signal.

 

Examples:

 

Ultrasound:

 

1 * 33 000 - 1 * 32 000 = 1 000 Hz

 

Audible range:

 

2 * 5 000 - 3 * 2 000 = 4 000 Hz

 

So at a system output present intermodulation products: 1 000 and 4 000 Hz.

 

When we cut ultrasound, there will: 1 000 Hz.

 

Levels of the products depend on non-linearity form.

 

Real spectrum of audio is almost solid. So number of intermodulation products are infinite too. They interfere with musical signal.

 

It is reason why impossibly to use musical signal to audio system estimation.

 

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5 hours ago, beerandmusic said:

nope what?

 

Start again from the beginning of the thread. All your questions have been answered, even if you didn't really like the answers because they didn't confirm your expectations.

"Science draws the wave, poetry fills it with water" Teixeira de Pascoaes

 

HQPlayer Desktop / Mac mini → Intona 7054 → RME ADI-2 DAC FS (DSD256)

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8 hours ago, firedog said:

None of that has anything to do with other factors that might influence listening SQ such as clean power. There are lots of things in audio that are debatable. This specific question and answer aren't one of them.

 

 So is the BS that several here appear to be pushing about 16/44.1 being all that's necessary for perfect stereo reproduction. 

 Incidentally, years later Elektor magazine published a design intended to correct the phase related problems of the original Sony players.

 

How a Digital Audio file sounds, or a Digital Video file looks, is governed to a large extent by the Power Supply area. All that Identical Checksums gives is the possibility of REGENERATING the file to close to that of the original file.

PROFILE UPDATED 13-11-2020

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7 minutes ago, sandyk said:

 

 So is the BS that several here appear to be pushing about 16/44.1 being all that's necessary for perfect stereo reproduction. 

 

Is anyone here actually saying this?

 

Beery seems to think this is what people are saying but I don't remember anyone actually saying this other than Beery himself.

Sometimes it's like someone took a knife, baby
Edgy and dull and cut a six inch valley
Through the middle of my skull

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4 hours ago, firedog said:

Yes, but at least on some recordings there could well be material above 44.1k, musical or otherwise. Recording at 96k or above eliminates this as a potential problem. 

 

Not quite , at least in the case of some of Barry Diament's recordings with genuine musical content to 57kHZ

 

How a Digital Audio file sounds, or a Digital Video file looks, is governed to a large extent by the Power Supply area. All that Identical Checksums gives is the possibility of REGENERATING the file to close to that of the original file.

PROFILE UPDATED 13-11-2020

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23 minutes ago, sandyk said:

 So is the BS that several here appear to be pushing about 16/44.1 being all that's necessary for perfect stereo reproduction. 

 Incidentally, years later Elektor magazine published a design intended to correct the phase related problems of the original Sony players.

The early Sony players multiplexed a single DAC for both channels resulting in one channel being delayed by 11.3 μs. That doesn't mean the format itself is insufficient.

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4 minutes ago, mansr said:

The early Sony players multiplexed a single DAC for both channels resulting in one channel being delayed by 11.3 μs. That doesn't mean the format itself is insufficient.

 My reply was a quick response to the statement below , and the earlier players such as the Sony CDP101 were quite fatiguing for long listening sessions .

Quote

......Philips and Sony really did think about these things......

 

 

How a Digital Audio file sounds, or a Digital Video file looks, is governed to a large extent by the Power Supply area. All that Identical Checksums gives is the possibility of REGENERATING the file to close to that of the original file.

PROFILE UPDATED 13-11-2020

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15 minutes ago, mansr said:

How do you know it's musical if you can't hear it?

 

There are many reports that the absence of the higher frequencies are noticed, even though we can't hear them directly with our ears.

 

How a Digital Audio file sounds, or a Digital Video file looks, is governed to a large extent by the Power Supply area. All that Identical Checksums gives is the possibility of REGENERATING the file to close to that of the original file.

PROFILE UPDATED 13-11-2020

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1 minute ago, Spacehound said:

They are not true.

 

 Bone conduction is one of the reported mechanisms for noticing HF above the normal hearing range.

 

How a Digital Audio file sounds, or a Digital Video file looks, is governed to a large extent by the Power Supply area. All that Identical Checksums gives is the possibility of REGENERATING the file to close to that of the original file.

PROFILE UPDATED 13-11-2020

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48 minutes ago, sandyk said:

 

Not quite , at least in the case of some of Barry Diament's recordings with genuine musical content to 57kHZ

 

I found his sample recordings quite bright, something which can easily be confirmed with a spectrum analyser.

 

 

 

sr005-01-2496.wav.png

"Science draws the wave, poetry fills it with water" Teixeira de Pascoaes

 

HQPlayer Desktop / Mac mini → Intona 7054 → RME ADI-2 DAC FS (DSD256)

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