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Interesting Papers re temporal resolution
John_Siau
post Jul 31 2009, 19:16
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From the first paragraph at: www.physics.sc.edu/kunchur/Acoustics-papers.htm
QUOTE
Our recent behavioral studies on human subjects proved that humans can discern timing alterations on a 5 microsecond time scale, indicating that that digital sampling rates used in common consumer audio (such as CD) are insufficient for fully preserving transparency.


This statement contains a glaring error:

44.1 kHz sampling PCM systems are perfectly capable of reproducing the phase of audible frequencies to picosecond accuracy. The need for 5-microsecond temporal accuracy does NOT indicate the need for a higher sample rate. It simply indicates that jitter must be less than 5-microseconds.


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John_Siau
post Jul 31 2009, 19:37
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From the summary paragraph in "Audibility of temporal smearing and time misalignment of acoustic signals" by Dr. Kunchur
QUOTE
These qualitative and anecdotal observations point to the possibility that human hearing may be sensitive to temporal errors, τ , that are shorter than the reciprocal of the limiting angular frequency [2πfmax]^−1 ≈ 9 μs, thus necessitating bandwidths in audio equipment that are much higher than fmax in order to preserve fidelity.


Again, the same glaring error:

The Nyquist frequency does NOT determine the temporal resolution of a PCM system. The entire premise of the paper is fundamentally flawed. The experiment proves the need for moderately controlled jitter but does not establish the need for higher sample rates.


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Canar
post Jul 31 2009, 20:12
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QUOTE (ncdrawl @ Jul 31 2009, 09:15) *
I have no problem complying with that, assuming that the SOP is enforced in an even-handed manner.
If you notice us not being even-handed, you are welcome to PM me personally, though there are certain technical points that are assumed to be true here on this forum. Asserting them usually does not invoke Term of Service #8, though asserting the contrary will. Nevertheless, you're welcome to assert the contrary if you have back-up and can make a solid case.


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Axon
post Jul 31 2009, 21:28
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QUOTE (John_Siau @ Jul 31 2009, 13:37) *
From the summary paragraph in "Audibility of temporal smearing and time misalignment of acoustic signals" by Dr. Kunchur
QUOTE
These qualitative and anecdotal observations point to the possibility that human hearing may be sensitive to temporal errors, τ , that are shorter than the reciprocal of the limiting angular frequency [2πfmax]^−1 ≈ 9 μs, thus necessitating bandwidths in audio equipment that are much higher than fmax in order to preserve fidelity.


Again, the same glaring error:

The Nyquist frequency does NOT determine the temporal resolution of a PCM system. The entire premise of the paper is fundamentally flawed. The experiment proves the need for moderately controlled jitter but does not establish the need for higher sample rates.


That seems too harsh to me - I agree that Dr. Kunchur is making huge conceptual mistakes with his treatment of digital audio, and in a lot of other things for that matter, but the experiment itself can't be dismissed so easily. Despite all the issues I have with the paper that I mentioned earlier, isn't it at least plausible on some sort of a priori level that, if one can guarantee a distortion of fairly reasonable consistency (internal to the ear), that testing of ultrasonics on that basis may be reasonable?
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andy_c
post Jul 31 2009, 21:30
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QUOTE (John_Siau @ Jul 31 2009, 12:16) *
This statement contains a glaring error:


Welcome to the hornets' nest smile.gif.

In all seriousness, he does attempt to address this in his FAQ document, in the last sentence of page 1 and the first couple of sentences of page 2. He states:

QUOTE
If there are two sharp peaks in sound pressure separated by 5 microseconds (which was the threshold upper bound determined in our experiments), they will merge together and the essential feature (the presence of two distinct peaks rather than one blurry blob) is destroyed. There is no ambiguity about this and no number of vertical bits or DSP can fix this. Hence the temporal resolution of the CD is inadequate for delivering the essence of the acoustic signal (2 distinct peaks).


Of course, you might wonder what the relationship of his test (audibility of a 5us time constant filtering a 7 kHz square wave) is to the "two distinct pulses separated by 5us" scenario. If so, you are not alone. However it does seem to bear a closer relationship to his experiment with the two loudspeaker drivers where the delay between them was varied to find the minimum detectable delay difference. But this has been criticized on the grounds that the spectrum of the close-together pulses vs. that of the pulses further separated can differ by an audible amount below 20 kHz.

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John_Siau
post Jul 31 2009, 22:15
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QUOTE (Axon @ Jul 31 2009, 16:28) *
That seems too harsh to me - I agree that Dr. Kunchur is making huge conceptual mistakes with his treatment of digital audio, and in a lot of other things for that matter, but the experiment itself can't be dismissed so easily. Despite all the issues I have with the paper that I mentioned earlier, isn't it at least plausible on some sort of a priori level that, if one can guarantee a distortion of fairly reasonable consistency (internal to the ear), that testing of ultrasonics on that basis may be reasonable?


I do not dismiss the test results that attempt to quantify our ability to hear abrupt temporal shifts. This is useful information. However, the paper assumes that these 5 usec temporal shifts cannot be represented in a 44.1 kHz PCM system. This assumption is simply false.


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WernerO
post Aug 1 2009, 06:46
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QUOTE (Axon @ Jul 31 2009, 21:28) *
Dr. Kunchur is... and in a lot of other things for that matter, but the experiment itself can't be dismissed so easily. ... isn't it at least plausible on some sort of a priori level that, if one can guarantee a distortion of fairly reasonable consistency (internal to the ear), that testing of ultrasonics on that basis may be reasonable?


I agree totally. The jumping to conclusions re digital audio is flawed, but that does not invalidate
both experiments, and there might be something interesting lurking there.


By the way:

QUOTE (Axon @ Jul 28 2009, 16:12) *
[*]When these issues are corrected, the values of ΔLp(2) for the RC-filter experiment fall from 1.4db down to the 0.2-0.3db range, and do not materially differ between the 3.9us and 4.7us cases. It therefore becomes extremely difficult to justify the results of the RC filter test due to nonlinear mixing.


Isn't that already in the RC paper, implicitly? One of the test subjects was tested to have hearing capability
to only less than 10kHz, while he faultlessly detected RCs down to 5.6us, and scored 8/10 on 4us (or whatever the figures were).


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2Bdecided
post Aug 1 2009, 08:24
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QUOTE (Axon @ Jul 31 2009, 21:28) *
That seems too harsh to me - I agree that Dr. Kunchur is making huge conceptual mistakes with his treatment of digital audio, and in a lot of other things for that matter, but the experiment itself can't be dismissed so easily. Despite all the issues I have with the paper that I mentioned earlier, isn't it at least plausible on some sort of a priori level that, if one can guarantee a distortion of fairly reasonable consistency (internal to the ear), that testing of ultrasonics on that basis may be reasonable?
Well, that's the whole point - but this flawed experiment didn't probe this properly - and all previous proper experiments show that, apart from via bone conduction, ultrasonics don't distort to create audible frequencies in the ear.

In equipment, yes, but not in the ear.

If someone proves otherwise (quite possible), it'll be interesting.

IIRC there was someone here who did (playing ultrasonic from a separate audio system(!) and ABXing presence / absence in the presence of an audible sound), but I didn't see how the thread ended.

Cheers,
David.
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rpp3po
post Aug 2 2009, 17:42
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I'm just coming back from vacation. Thanks for the many insightful posts, I have just learned a lot! Until now I always thought that PCM being bandwidth limited and requiring low pass filtering is just about frequencies that no human ear would care about anyway and no big deal else.

Comprehending that the plot of a properly encoded PCM bitstream of a square wave is not square, why low passing before ADC is so important, and why perfect square waves don't fit into PCM in theory took me a big leap forward.

I had already checked DVD-A prices for my favorite records online... happy.gif But until I see Kunchur refuting some of this thread's objections I'll stay as happy with Redbook as I always have been.

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Arnold B. Kruege...
post Aug 3 2009, 01:24
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QUOTE (rpp3po @ Aug 2 2009, 12:42) *
But until I see Kunchur refuting some of this thread's objections I'll stay as happy with Redbook as I always have been.


As you seem to be saying, there's a lot of interesting twists along the way, but getting back to basics is very often a good thing.

Setting up a proper listening test that shows that high sample rates do make a difference with music is simpler and less costly than ever.

There is now a ton of high sample rate audio kicking around for free or not much money. Actually setting up the experiment can be accomplished with not only reasonably-priced buyware, but even with freeware.

There seems to be only one minor thing lacking - a pair of ears that produce results that are positive for high sample rates.

Whenever we see a paper like Kunchur's, we need to get back to basics. The basics are that just about anybody who wants to can set up a reliable listening test involving readily-available musical program material that has the potential to produce a positive outcome. Nobody seems to be able to do it. Instead, all we get are these abstract tests like Kunchur's, involving some most definately non-musical waveforms with questionable relevance and inconclusive results.
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rpp3po
post Aug 3 2009, 01:42
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QUOTE (Arnold B. Krueger @ Aug 3 2009, 02:24) *
Nobody seems to be able to do it. Instead, all we get are these abstract tests like Kunchur's, involving some most definately non-musical waveforms with questionable relevance and inconclusive results.


I agree. Not a single positive result in a decade is simple but mortgageable evidence.

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ncdrawl
post Aug 3 2009, 04:59
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QUOTE (rpp3po @ Aug 2 2009, 20:42) *
QUOTE (Arnold B. Krueger @ Aug 3 2009, 02:24) *
Nobody seems to be able to do it. Instead, all we get are these abstract tests like Kunchur's, involving some most definately non-musical waveforms with questionable relevance and inconclusive results.


I agree. Not a single positive result in a decade is simple but mortgageable evidence.



Dr. Kunchur will be posting confutations in form of FAQ updates...will be doing that regularly.

Apparently he has been bestowed with a great deal of common sense to go with his impressive educational credentials... IE he avoids online arguments. Very smart decision.



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krabapple
post Aug 3 2009, 07:57
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QUOTE (ncdrawl @ Aug 2 2009, 23:59) *
QUOTE (rpp3po @ Aug 2 2009, 20:42) *
QUOTE (Arnold B. Krueger @ Aug 3 2009, 02:24) *
Nobody seems to be able to do it. Instead, all we get are these abstract tests like Kunchur's, involving some most definately non-musical waveforms with questionable relevance and inconclusive results.


I agree. Not a single positive result in a decade is simple but mortgageable evidence.



Dr. Kunchur will be posting confutations in form of FAQ updates...will be doing that regularly.


So, when's the next regular installment?


QUOTE
Apparently he has been bestowed with a great deal of common sense to go with his impressive educational credentials...


rolleyes.gif

QUOTE
IE he avoids online arguments. Very smart decision.



Possibly, but perhaps not for the reason you think. The closer to 'real time' an interrogation by knowledgeable critics is, the more stressful it can be. In a FAQ one is in total control of which questions will be addressed and how in-depth the address will be. On a forum you don't have that control.

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WernerO
post Aug 3 2009, 08:52
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QUOTE (ncdrawl @ Aug 3 2009, 04:59) *
Dr. Kunchur will be posting confutations in form of FAQ updates...will be doing that regularly.


Might we ask then that he keeps/publishes a record of the changes, so that
we don't have to re-read all all of the time?
Further, there already is something interesting in an earlier
version of the FAQ document that is missing in the present one, if I remember
correctly.

It had to do with the time accuracy of 44.1kHz sampled systems ;-)

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honestguv
post Aug 3 2009, 08:54
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> Dr. Kunchur will be posting confutations in form of FAQ updates...will be
> doing that regularly.

That will be interesting to see if he keeps digging himself deeper into a hole or starts trying to climb out. Unless he is a completely lost in audiophile belief (he is clearly a believer) he must have twigged by now that his lack of understanding of the information contained in a set of samples has lead him to make a bit of a fool of himself among his peers. But he is getting attention from the audiophile community. So which is more important to him? We will see.

> Apparently he has been bestowed with a great deal of common sense to go with
> his impressive educational credentials...

Why do you find them impressive? If you find scientific credentials impressive, why do you discard the bulk of relevant scientific work on sound perception performed by people with a higher standing in the field than your good Dr.?

> he avoids online arguments. Very smart decision.

In his position it would be rather unwise to do otherwise.
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andy_c
post Aug 3 2009, 17:34
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Regarding my earlier post about the lack of second harmonic data in the "Temporal resolution of hearing probed by bandwidth restriction" article, it looks like he published those data in another article, "Probing the temporal resolution and bandwidth of human hearing" (PDF file here). I had previously used the number 1e-3 for the second harmonic relative to the fundamental, read from his graph in figure 4 of "Temporal resolution of hearing probed by bandwidth restriction", to compute the required accuracy for the duty cycle of the square wave. His actual tabular data show a second harmonic relative to the fundamental of 0.0003, or -70.4 dB, rather than -60 dB as I had originally read from his graph. Plugging in the numbers, the previous requirement for the duty cycle accuracy, which was 50 +/- 0.032 percent, now becomes 50 +/- 0.0095 percent for a -70.4 dB second harmonic. That assumes zero second-order distortion in the buffer that follows the square wave generator, the buffer after the LPF, the headphone amplifier, the headphones themselves, the microphone, the mic preamp, and the 12-bit A/D converter in the oscilloscope used to digitize the data. I've worked with test equipment for many years, and in all that time I've never seen a square wave generator specified for its second harmonic content. The specifications for the pulse generator he used certainly don't show anything like that. I suppose it would be possible to put it in the variable pulse width mode and adjust its parameters to minimize the second harmonic, but according to Axon's communications with Dr. Kunchur, the spectrum computation takes days(?). That would make such an approach impractical. Or it could actually be that the square wave symmetry of the generator is that good. It seems unlikely though.

On the plus side, as long as this ratio were low enough, and constant with varying bandwidth, it probably wouldn't influence the results at all. The very low values of second harmonic just jumped out at me, considering how tiny errors in duty cycle of the square wave (and other contributors to errors in half-wave symmetry) result in non-negligible second harmonic spectral content, as well as how the actual data for the second harmonic depend on unspecified parameters of the square wave generator. Also, the second harmonic data in figure 4 of "Temporal resolution of hearing probed by bandwidth restriction" looks like a noise blip, rather than having a wide baseline as the other harmonics do - which also looks odd. I suppose this could be due to the resolution limit of the 12-bit converter used in the scope.

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Axon
post Aug 3 2009, 19:02
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QUOTE (WernerO @ Aug 3 2009, 02:52) *
QUOTE (ncdrawl @ Aug 3 2009, 04:59) *
Dr. Kunchur will be posting confutations in form of FAQ updates...will be doing that regularly.


Might we ask then that he keeps/publishes a record of the changes, so that
we don't have to re-read all all of the time?
Further, there already is something interesting in an earlier
version of the FAQ document that is missing in the present one, if I remember
correctly.

It had to do with the time accuracy of 44.1kHz sampled systems ;-)


Oh, man. I completely forgot about that. I've noticed the same thing. I have a printout of his original FAQ, so at least we'll know exactly what he's changing.

BTW, I find this behavior highly unprofessional. I would almost find it tantamount to lying.

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honestguv
post Aug 3 2009, 19:35
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QUOTE (Axon @ Aug 3 2009, 20:02) *
BTW, I find this behavior highly unprofessional. I would almost find it tantamount to lying.

Little sympathy for a man with problems? His beliefs are not holding up, the basis for his audiophile fame is starting to crumble, he has published papers in his name with erroneous content, how to fix the obvious false statements in his FAQ but still give audiophiles what they want to see?, should he attempt to keep going? read a bit about sampling and then correct the mistakes? delete the FAQ, remove the links to the hardcopies of the papers and keep quiet for a bit,...

It is interesting to see what he will do and if he ends up trusting existing scientific knowledge a bit more and audiophile beliefs a bit less.
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benski
post Aug 3 2009, 20:10
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I need to do further reading to do a full refutation, but I think I figured out where the problem is right.

Dr Kunchur shows that humans can discern temporal resolution down to 5 microseconds. I mostly agree with this - it's a fundamental part of pitch perception. Where he makes the mistake is assuming that 44,100Hz sampling rate only has a temporal resolution of 22 microseconds. This is patently false. 44100Hz audio has far more temporal resolution than this, due to the antialiasing filter. If you model the impulse response function of the antialiasing function as a sinc wave, small (<20microseconds) delays in an impulse event will correspond to vastly different outputs from the sinc() function. If the input impulse is perfectly aligned, the sinc() output will give 1 followed by an infinite amount of zeroes, but a half-sample (11 microsecond) delay will give 2/PI followed by a series of decaying values. The reconstruction filter will output the original half-sample delayed signal, but in bandlimited form. The half-second delay on the impulse will still be present on the output signal.

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benski
post Aug 3 2009, 20:24
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Also, for his FAQ example of two peaks separated by 5 microseconds, my statement still holds true. The two peaks will "ring" the anti-alias filter in such a way that those two peaks still have an impact into the digital sampling (and on the eventual post-reconstruction-filter output). The waveform will look different and the high frequency spectra will be lost but the "information" of the two separate peaks will remain in the final analog output.
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WernerO
post Aug 3 2009, 20:56
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QUOTE (benski @ Aug 3 2009, 21:24) *
two peaks separated by 5 microseconds, my statement still holds true. ...The waveform will look different and the high frequency spectra will be lost but the "information" of the two separate peaks will remain in the final analog output.


Er, no. They would be merged into one peak. Do the experiment.

But this of course does not prove that 44.1kHz is insufficient because of the simple
fact that it has not been proven that an audio system can only be audibly transparent
when it keeps 5us impulses separated. That's the error.

It's a bit like stating that airplanes are rubbish because you can't do space missions
with them.


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Axon
post Aug 3 2009, 21:04
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QUOTE (WernerO @ Aug 3 2009, 14:56) *
QUOTE (benski @ Aug 3 2009, 21:24) *
two peaks separated by 5 microseconds, my statement still holds true. ...The waveform will look different and the high frequency spectra will be lost but the "information" of the two separate peaks will remain in the final analog output.


Er, no. They would be merged into one peak. Do the experiment.


They're still separate in the frequency domain.
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benski
post Aug 3 2009, 21:22
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QUOTE (WernerO @ Aug 3 2009, 15:56) *
QUOTE (benski @ Aug 3 2009, 21:24) *
two peaks separated by 5 microseconds, my statement still holds true. ...The waveform will look different and the high frequency spectra will be lost but the "information" of the two separate peaks will remain in the final analog output.


Er, no. They would be merged into one peak. Do the experiment.

But this of course does not prove that 44.1kHz is insufficient because of the simple
fact that it has not been proven that an audio system can only be audibly transparent
when it keeps 5us impulses separated. That's the error.

It's a bit like stating that airplanes are rubbish because you can't do space missions
with them.


Please re-read the part I just bolded in the "quote" block above.
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honestguv
post Aug 3 2009, 21:47
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QUOTE (WernerO @ Aug 3 2009, 21:56) *
But this of course does not prove that 44.1kHz is insufficient because of the simple
fact that it has not been proven that an audio system can only be audibly transparent
when it keeps 5us impulses separated. That's the error.

What do these two peaks in the time domain represent? Without a formulation of a comprehensible hypothesis how can one perform a test and claim an error? (Ditto for benski.)
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benski
post Aug 3 2009, 22:04
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QUOTE (honestguv @ Aug 3 2009, 16:47) *
QUOTE (WernerO @ Aug 3 2009, 21:56) *
But this of course does not prove that 44.1kHz is insufficient because of the simple
fact that it has not been proven that an audio system can only be audibly transparent
when it keeps 5us impulses separated. That's the error.

What do these two peaks in the time domain represent? Without a formulation of a comprehensible hypothesis how can one perform a test and claim an error? (Ditto for benski.)


They are supposed to repesentation two sounds occuring nearly, but not quite, simultaneously, for example two drums where the mallets hit 5 microseconds apart.

The proper way to test would be digitize the sound @ 44.1khz and again at something much higher like 384khz (which has a sampling period smaller than 5 microseconds), and do a blind ABX test with a set of listeners (using the same criteria for choosing listeners as the experiment in question). Most of us at HA know better than to trust a spectrogram. Although my point was simply that the post-experiment statement of the researcher can be easily disproven using Shannon's sampling theorem.

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