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CHORD M-Scaler Review (Upsampler)

Rate this product:

  • 1. Poor (headless panther)

    Votes: 328 89.9%
  • 2. Not terrible (postman panther)

    Votes: 9 2.5%
  • 3. Fine (happy panther

    Votes: 7 1.9%
  • 4. Great (golfing panther)

    Votes: 21 5.8%

  • Total voters
    365

Snoopy

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the M-Scaler outputs 798Khz. not by adding information that wasn't there. but by doing a job of making the input signal closer to an analogue wave sign for the DAC to output for the pre-amp to amplify so the signal sounds better. I think it does. whatever you all say if you blindfolded me and took it away I would know instantly. there's a button to change the sample rate on the M-Scaler. can easily tell the difference between max scaling and no scaling.
You could do this in software like HQ player or roon... 1536 khz pcm or DSD1024.

the M-scaler is just a waste of space
 

HP9000

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No claim is made that higher-ranked products based on distortion (SINAD actually) sound different than lower-ranked ones, if the lower-ranked ones already exceed the limits of audibility. So your analogy doesn’t support the point you’re trying to make.
The lower ranked products having no sound difference from the higher ranked ones is only true if all of the metrics are below audible threshold, and how can we confirm that they measure below audibility if some of the measurements (time response) aren't shown or taken?

(Edit: and my point is that time resolution not needing improvement is not a reason for it to be excluded from reviews when other metrics that don't need improvement are still included)
 
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tmtomh

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The lower ranked products having no sound difference from the higher ranked ones is only true if all of the metrics are below audible threshold, and how can we confirm that they measure below audibility if some of the measurements (time response) aren't shown or taken?

(Edit: and my point is that time resolution not needing improvement is not a reason for it to be excluded from reviews when other metrics that don't need improvement are still included)

@amirm and others are more knowledgeable than I am and can give a more complete answer, but time resolution is not the same as distortion and noise. The “time resolution” is about sample rates and clocking accuracy, and is a solved problem with DACs. An audible lack of “time resolution” manifests itself as digital clicks, dropouts, etc.
 
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amirm

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Many of the products you measure/review have many of their distortion figures well below audible threshold, yet they are included and ranked by their slight differences. Why is time response in particular excluded?
For the same reason that the manufacturer/designer gives you nothing in that regard. There is no there, there.

 

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voodooless

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People don't hear square waves the way they look because our hearing is limited to 20kHz.
Besides, we have very good indicators or performance: the multitone and THD+N vs frequency. If it were not “fast” enough, you’d have high frequency distortion.
 

Geert

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Besides, we have very good indicators or performance: the multitone and THD+N vs frequency. If it were not “fast” enough, you’d have high frequency distortion.

Exactly, these measurements show you what matters. The time domain is an audiophile labyrinth.
 

HP9000

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@tmtohm "sample rates and clocking accuracy... digital clicks, dropouts, etc."
@amirm "Does Phase Distortion/Shift Matter in Audio? (no*)"

Sample rates, clocks, phase.... I'm talking more specifically about the square wave response, soon any further comments or replies I post on this subject will be posted in the appropriate thread seeing there is one and a video on it: https://www.audiosciencereview.com/...sting-of-audio-products-video-tutorial.20984/

@Geert "People don't hear square waves the way they look because our hearing is limited to 20kHz"

Our tonal sensitivity is limited to around 20khz, and our sensitivity to amplitude changes in time - of a given audible (fundamental) frequency and a given audible difference in amplitude - is limited to an equivalent of 100khz, which is 5 times as high.

@voodooless "Besides, we have very good indicators for performance: the multitone and THD+N vs frequency. If it were not “fast” enough, you’d have high frequency distortion."
@Geert "Exactly, these measurements show you what matters."

A device can have superior time performance and inferior multitone and THD+N, and vice versa. The Rehdeko speaker has very good timing measurements, and given they are a crossover-less speaker with limited surface area they probably have higher intermodulation. Filter-less NOS dacs have superior timing performance than delta-sigma and have higher intermodulation and probably also THD+N. This proves the point that a time measurement like a step or impulse response shows exclusive results that can't necessarily be inferred from other measurements, and so shouldn't be excluded from reviews.
 

Axo1989

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@tmtohm "sample rates and clocking accuracy... digital clicks, dropouts, etc."
@amirm "Does Phase Distortion/Shift Matter in Audio? (no*)"

Sample rates, clocks, phase.... I'm talking more specifically about the square wave response, soon any further comments or replies I post on this subject will be posted in the appropriate thread seeing there is one and a video on it: https://www.audiosciencereview.com/...sting-of-audio-products-video-tutorial.20984/

@Geert "People don't hear square waves the way they look because our hearing is limited to 20kHz"

Our tonal sensitivity is limited to around 20khz, and our sensitivity to amplitude changes in time - of a given audible (fundamental) frequency and a given audible difference in amplitude - is limited to an equivalent of 100khz, which is 5 times as high.

@voodooless "Besides, we have very good indicators for performance: the multitone and THD+N vs frequency. If it were not “fast” enough, you’d have high frequency distortion."
@Geert "Exactly, these measurements show you what matters."

A device can have superior time performance and inferior multitone and THD+N, and vice versa. The Rehdeko speaker has very good timing measurements, and given they are a crossover-less speaker with limited surface area they probably have higher intermodulation. Filter-less NOS dacs have superior timing performance than delta-sigma and have higher intermodulation and probably also THD+N. This proves the point that a time measurement like a step or impulse response shows exclusive results that can't necessarily be inferred from other measurements, and so shouldn't be excluded from reviews.

Interesting.
 

tmtomh

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@tmtohm "sample rates and clocking accuracy... digital clicks, dropouts, etc."
@amirm "Does Phase Distortion/Shift Matter in Audio? (no*)"

Sample rates, clocks, phase.... I'm talking more specifically about the square wave response, soon any further comments or replies I post on this subject will be posted in the appropriate thread seeing there is one and a video on it: https://www.audiosciencereview.com/...sting-of-audio-products-video-tutorial.20984/

@Geert "People don't hear square waves the way they look because our hearing is limited to 20kHz"

Our tonal sensitivity is limited to around 20khz, and our sensitivity to amplitude changes in time - of a given audible (fundamental) frequency and a given audible difference in amplitude - is limited to an equivalent of 100khz, which is 5 times as high.

@voodooless "Besides, we have very good indicators for performance: the multitone and THD+N vs frequency. If it were not “fast” enough, you’d have high frequency distortion."
@Geert "Exactly, these measurements show you what matters."

A device can have superior time performance and inferior multitone and THD+N, and vice versa. The Rehdeko speaker has very good timing measurements, and given they are a crossover-less speaker with limited surface area they probably have higher intermodulation. Filter-less NOS dacs have superior timing performance than delta-sigma and have higher intermodulation and probably also THD+N. This proves the point that a time measurement like a step or impulse response shows exclusive results that can't necessarily be inferred from other measurements, and so shouldn't be excluded from reviews.

You're still not defining "time performance" in a way that makes any sense. You seem to be saying that we can detect changes in amplitude with a speed that exceeds the speed of the wavelength of a 20kHz sound. But that doesn't mean that an audio reproduction device needs to be able to reproduce 100kHz sounds - nor does it mean that there are digital devices that aren't "fast enough" to accurately reproduce amplitude changes in recorded music. If that were true, then digital sampling theory would be invalid and plenty of gear and systems - way beyond just hi-fi equipment - simply would not work.
 

voodooless

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A device can have superior time performance and inferior multitone and THD+N, and vice versa. The Rehdeko speaker has very good timing measurements, and given they are a crossover-less speaker with limited surface area they probably have higher intermodulation.
That’s not a DAC. A speakers distortion profile and source is totally different from a DAC, and the magnitude is orders or magnitude bigger. You can’t just make that compassion.
Filter-less NOS dacs have superior timing performance than delta-sigma and have higher intermodulation and probably also THD+N.
They don’t! They can’t reproduce a square wave offset between samples very well (or high frequency content for that matter). Your conflating being able to draw pretty square waves with timing performance. The fact that they have high distortion means they cannot reproduce the origin signal very well. How can something that is worse at reproduction the original signal have better timing performance?

This proves the point that a time measurement like a step or impulse response shows exclusive results that can't necessarily be inferred from other measurements, and so shouldn't be excluded from reviews.
Impulse response can be converted to frequency and phase, and vice versa. They are two sides of the same coin.
 
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Geert

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Our tonal sensitivity is limited to around 20khz, and our sensitivity to amplitude changes in time - of a given audible (fundamental) frequency and a given audible difference in amplitude - is limited to an equivalent of 100khz, which is 5 times as high.

Reference?
 

HP9000

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@DonR, @tmtomh, @voodooless, @Geert

"Impulse response can be converted to frequency and phase, and vice versa. They are two sides of the same coin."
Two devices having linear frequency response from 20hz to 20khz can show different time responses in the way of their impulse/square wave throughout that bandwidth, so having only the frequency-magnitude chart is going to be insufficient.

"They can’t reproduce a square wave offset between samples very well"
That is an issue of quantization/sample rate limitations.

"They don’t!"
A delta-sigma's innate use of oversampling affects the impulse response and puts it at a disadvantage in comparison to a filter-less non-oversampling dac's impulse response. A dac may have low sample rate music or oversampled music played into it and may have a reconstruction lpf (all affecting the impulse response), but these are not innate qualities of say a resistor ladder architecture, hence my use of a filter-less NOS kind for emphasis.

"(or high frequency content for that matter)."
A filter-less dac is going to have an abundance of high frequency content. It is supposedly as a result of limited bandwidth of upstream devices that this would result in intermodulation, but if intermodulation is understood as distortion in the form of frequencies that weren't in the original signal, then all that would really be the cause is the modulation frequency (ex. 44.1khz). I suspect that when intermodulation tests are taken on a dac that it is from 20hz to 20khz, so the program does not recognize/is not informed of the higher-than-20khz sample rate.
Hearing 44.1khz or 48khz music played through a filter-less NOS, people have both good and bad to say about it; it is realistic, but also 'hazy' sounding. This is because the amplitude changes that are recreating audible tones are at a rate that is less than half of our time sensitivity, yet 44,100 times per sec is still a high rate generally speaking (hence 'hazy'). This is why non-oversampled 96khz (near the aforementioned 100khz limit) music played through the dac would overcome the hazy sound and the need for a reconstruction lpf and oversampling (read delta-sigma aswell).

"The fact that they have high distortion means they cannot reproduce the origin signal very well. How can something that is worse at reproduction the original signal have better timing performance?"
A device can have high distortion in some areas and low distortion in others, i.e. something can be worse at reproducing other aspects of a signal and better at the timing aspect.

"That’s not a DAC."
That's just my point, because the idea that you can't necessarily infer from positive multitone and THD+N vs frequency that the impulse/square wave will also show positive applies to not just dacs but amps and speakers, meaning more measurements have to be shown to draw conclusions about sound quality.

"But that doesn't mean that an audio reproduction device needs to be able to reproduce 100kHz sounds - nor does it mean that there are digital devices that aren't "fast enough" to accurately reproduce amplitude changes in recorded music. If that were true, then digital sampling theory would be invalid and plenty of gear and systems - way beyond just hi-fi equipment - simply would not work."
First, given frequency is a continuum, I'd be surprised if most audio electronics are incapable of playing 100khz, instead of just playing it at a very low level. Second, of course a device is not broken if it can't play linearly up to 100khz or more, it is to say that there is room for improvement.

"Phase distortion above the audio band does not matter: https://www.audiosciencereview.com/...se-distortion-shift-matter-in-audio-no.24026/"
"You seem to be saying that we can detect changes in amplitude with a speed that exceeds the speed of the wavelength of a 20kHz sound."
"Reference?"

At 13:05 in the video by amirm that is linked he says that time differential is a main factor in localization, and we can ask: what amount of time?
My previous post stating a 100khz-equivalent time sensitivity was gotten by a quick search while making the post, the context of which is even the same as in amirm's video: "One final fascinating phenomenon is the interaction of the two ears in interpreting sound. A big advantage to having two ears is the ability to accurately localize sound. There are two cues that the brain uses in doing this. The first is, as we mentioned earlier, the fact that the time of arrival of sounds in the two ears is slightly different. The closer ear receives the sound slightly earlier. The brain is sensitive to differences in time of arrival of as small as 10 microseconds, and can use this to pinpoint the location of the sound. The second cue is the fact that sounds arrive at slightly different amplitudes in the two ears. This is because the head causes an acoustic shadow which attenuates a sound coming from the opposite direction. By comparing the amplitude of the sounds in the two ears, the location of the source can be identified.

But if this is true, then how do we localize indoors? Sounds bounce numerous times off walls, ceilings, floors and other objects which would totally confuse the brain. It turns out that there is a precedence effect by which the brain only pays attention to the first wavefront that reaches it. All the subsequent echoes are ignored for the purpose of localization."

(From: https://web.mit.edu/2.972/www/reports/ear/ear.html)

What is interesting is that I wasn't even going to bother making a search to check and was going to put the same number from memory because I had read on this subject prior and from other sources. They are consistent with each other. I will find the other website I came across, it is somewhere in my phone's files.
 
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Sir Sanders Zingmore

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@DonR, @tmtohm, @voodooless, @Geert


"Impulse response can be converted to frequency and phase, and vice versa. They are two sides of the same coin."
Two devices having linear frequency response from 20hz to 20khz can show different time responses in the way of their impulse/square wave throughout that bandwidth, so having only the frequency-magnitude chart is going to be insufficient.

"They can’t reproduce a square wave offset between samples very well"
That is an issue of quantization/sample rate limitations.

"They don’t!"
A delta-sigma's innate use of oversampling affects the impulse response and puts it at a disadvantage in comparison to a filter-less non-oversampling dac's impulse response. A dac may have low sample rate music or oversampled music played into it and may have a reconstruction lpf (all affecting the impulse response), but these are not innate qualities of say a resistor ladder architecture, hence my use of a filter-less NOS kind for emphasis.

"(or high frequency content for that matter)."
A filter-less dac is going to have an abundance of high frequency content. It is supposedly as a result of limited bandwidth of upstream devices that this would result in intermodulation, but if intermodulation is understood as distortion in the form of frequencies that weren't in the original signal, then all that would really be the cause is the modulation frequency (ex. 44.1khz). I suspect that when intermodulation tests are taken on a dac that it is from 20hz to 20khz, so the program does not recognize/is not informed of the higher-than-20khz sample rate.
Hearing 44.1khz or 48khz music played through a filter-less NOS, people have both good and bad to say about it; it is realistic, but also 'hazy' sounding. This is because the amplitude changes that are recreating audible tones are at a rate that is less than half of our time sensitivity, yet 44,100 times per sec is still a high rate generally speaking (hence 'hazy'). This is why non-oversampled 96khz (near the aforementioned 100khz limit) music played through the dac would overcome the hazy sound and the need for a reconstruction lpf and oversampling (read delta-sigma aswell).

"The fact that they have high distortion means they cannot reproduce the origin signal very well. How can something that is worse at reproduction the original signal have better timing performance?"
A device can have high distortion in some areas and low distortion in others, i.e. something can be worse at reproducing other aspects of a signal and better at the timing aspect.

"That’s not a DAC."
That's just my point, because the idea that you can't necessarily infer from positive multitone and THD+N vs frequency that the impulse/square wave will also show positive applies to not just dacs but amps and speakers, meaning more measurements have to be shown to draw conclusions about sound quality.

"But that doesn't mean that an audio reproduction device needs to be able to reproduce 100kHz sounds - nor does it mean that there are digital devices that aren't "fast enough" to accurately reproduce amplitude changes in recorded music. If that were true, then digital sampling theory would be invalid and plenty of gear and systems - way beyond just hi-fi equipment - simply would not work."
First, given frequency is a continuum, I'd be surprised if most audio electronics are incapable of playing 100khz, instead of just playing it at a very low level. Second, of course a device is not broken if it can't play linearly up to 100khz or more, it is to say that there is room for improvement.

"Phase distortion above the audio band does not matter: https://www.audiosciencereview.com/...se-distortion-shift-matter-in-audio-no.24026/"
"You seem to be saying that we can detect changes in amplitude with a speed that exceeds the speed of the wavelength of a 20kHz sound."
"Reference?"

At 13:05 in the video by amirm that is linked he says that time differential is a main factor in localization, and we can ask: what amount of time?
My previous post stating a 100khz-equivalent time sensitivity was gotten by a quick search while making the post, the context of which is even the same as in amirm's video: "One final fascinating phenomenon is the interaction of the two ears in interpreting sound. A big advantage to having two ears is the ability to accurately localize sound. There are two cues that the brain uses in doing this. The first is, as we mentioned earlier, the fact that the time of arrival of sounds in the two ears is slightly different. The closer ear receives the sound slightly earlier. The brain is sensitive to differences in time of arrival of as small as 10 microseconds, and can use this to pinpoint the location of the sound. The second cue is the fact that sounds arrive at slightly different amplitudes in the two ears. This is because the head causes an acoustic shadow which attenuates a sound coming from the opposite direction. By comparing the amplitude of the sounds in the two ears, the location of the source can be identified.

But if this is true, then how do we localize indoors? Sounds bounce numerous times off walls, ceilings, floors and other objects which would totally confuse the brain. It turns out that there is a precedence effect by which the brain only pays attention to the first wavefront that reaches it. All the subsequent echoes are ignored for the purpose of localization."

(From: https://web.mit.edu/2.972/www/reports/ear/ear.html)

What is interesting is that I wasn't even going to bother making a search to check and was going to put the same number from memory because I had read on this subject prior and from other sources. They are consistent with each other. I will find the other website I came across, it is somewhere in my phone's files.
Fascinating. You have lots and lots of reasons why this device should make an audible difference, and yet it doesn’t.
 

voodooless

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Impulse response can be converted to frequency and phase, and vice versa. They are two sides of the same coin.
Two devices having linear frequency response from 20hz to 20khz can show different time responses in the way of their impulse/square wave throughout that bandwidth, so having only the frequency-magnitude chart is going to be insufficient.
That's why you need magnitude AND phase, as I mentioned.
"They can’t reproduce a square wave offset between samples very well"
That is an issue of quantization/sample rate limitations.
So we need oversampling after all? Obviously, if you shift all the issues high enough into the ultrasonic, they become less problematic. It's like cleaning your room by shoving all the shit you have lying around into a far away closet, so nobody sees. It's still there though..
"They don’t!"
A delta-sigma's innate use of oversampling affects the impulse response and puts it at a disadvantage in comparison to a filter-less non-oversampling dac's impulse response.
We don't listen to an impulse response. We listen to actual music, and an oversampling DAC can and will reproduce the original waveform better than your NOS DAC, regardless of the sample rate. There is no disadvantage.
"(or high frequency content for that matter)."
A filter-less dac is going to have an abundance of high frequency content. It is supposedly as a result of limited bandwidth of upstream devices that this would result in intermodulation
No, the intermodulation is not generated upstream, it's directly output by the NOS DAC. If you mean intermodulation products that reflect back to the audible part: obviously they are there, because a NOS cannot properly reconstruct the original waveform.
but if intermodulation is understood as distortion in the form of frequencies that weren't in the original signal, then all that would really be the cause is the modulation frequency (ex. 44.1khz).
What?
I suspect that when intermodulation tests are taken on a dac that it is from 20hz to 20khz, so the program does not recognize/is not informed of the higher-than-20khz sample rate.
I have no idea what you mean?
Hearing 44.1khz or 48khz music played through a filter-less NOS, people have both good and bad to say about it; it is realistic, but also 'hazy' sounding. This is because the amplitude changes that are recreating audible tones are at a rate that is less than half of our time sensitivity, yet 44,100 times per sec is still a high rate generally speaking (hence 'hazy'). This is why non-oversampled 96khz (near the aforementioned 100khz limit) music played through the dac would overcome the hazy sound and the need for a reconstruction lpf and oversampling (read delta-sigma aswell).
No, it just shoves the shit up in frequency, making it less audible.
"The fact that they have high distortion means they cannot reproduce the origin signal very well. How can something that is worse at reproduction the original signal have better timing performance?"
A device can have high distortion in some areas and low distortion in others, i.e. something can be worse at reproducing other aspects of a signal and better at the timing aspect.
Distortion is distortion.. We can measure it, and it covers all kinds of distortion, also in timing.
My previous post stating a 100khz-equivalent time sensitivity
Sample rate != time resolution. As long as you think it is, you're lost...
 

Geert

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At 13:05 in the video by amirm that is linked he says that time differential is a main factor in localization, and we can ask: what amount of time? My previous post stating a 100khz-equivalent time sensitivity was gotten by a quick search while making the post, the context of which is even the same as in amirm's video: "One final fascinating phenomenon is the interaction of the two ears in interpreting sound. A big advantage to having two ears is the ability to accurately localize sound. There are two cues that the brain uses in doing this. The first is, as we mentioned earlier, the fact that the time of arrival of sounds in the two ears is slightly different. The closer ear receives the sound slightly earlier. The brain is sensitive to differences in time of arrival of as small as 10 microseconds, and can use this to pinpoint the location of the sound.

You're now referring to interaural time difference (ITD), where for low frequency test signals the upper sensitivity threshold is indeed 10 μs (Yost W. - Discriminations of interaural phase differences. J. Acoust. Soc. Am, 1974.). However, it doesn't make any sense to translate this time value into a frequency (the 100 kHz you referred to) and suggest the Fourier transfer of a square wave is incomplete. It's like saying I had lunch at 0,083 Hz.

Also, ITD refers to the difference in arrival time of an angled sound source between the 2 ears as a result of different path lengths. It's not a value that indicates absolute time resolution sensitivity, like an impulse being early without a difference in path length. This kind of timing is being researched in studies about auditory temporal resolution or perception. For normal hearing and low frequencies the time resolution is 3 ms at best ('Temporal Resolution in Normal-Hearing and Hearing-Impaired Listeners Using Frequency-Modulated Stimuli', John P. Madden, Lawrence Feth, 1992). That's 130 times more than the period of a 44.1 kHz sample rate. Not that it really matters, because digital audio timing resolution isn't limited by sample frequency (because we listen to the reconstructed signal).
 

tmtomh

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You're now referring to interaural time difference (ITD), where for low frequency test signals the upper sensitivity threshold is indeed 10 μs (Yost W. - Discriminations of interaural phase differences. J. Acoust. Soc. Am, 1974.). However, it doesn't make any sense to translate this time value into a frequency (the 100 kHz you referred to) and suggest the Fourier transfer of a square wave is incomplete. It's like saying I had lunch at 0,083 Hz.

Also, ITD refers to the difference in arrival time of an angled sound source between the 2 ears as a result of different path lengths. It's not a value that indicates absolute time resolution sensitivity, like an impulse being early without a difference in path length. This kind of timing is being researched in studies about auditory temporal resolution or perception. For normal hearing and low frequencies the time resolution is 3 ms at best ('Temporal Resolution in Normal-Hearing and Hearing-Impaired Listeners Using Frequency-Modulated Stimuli', John P. Madden, Lawrence Feth, 1992). That's 130 times more than the period of a 44.1 kHz sample rate. Not that it really matters, because digital audio timing resolution isn't limited by sample frequency (because we listen to the reconstructed signal).

Great explanation - thanks!

And your last sentence is so key - it's what folks seem to persistently forget or neglect when they make these "time resolution" claims. It always strikes me how much incorrect info about digital audio can be traced back to the stair-step fallacy.
 
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JSmith

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It's like saying I had lunch at 0,083 Hz.
It's quite a good restaurant I hear too... although timing is quite important there, as in one must book in advance and there is strict table duration. Numbers are finite and the tables are not modular. The food is always on schedule, the gourmet chef ensures the timing is resolute. One must sample each item on the menu to understand the intricate nuance of the flavours... of course some excess sampling may be required for certain dishes. It really is on a grand scale.


JSmith
 
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