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"Though our tests failed to substantiate the claimed advantages of high-resolution encoding for two-channel audio, one trend became obvious very quickly and held up throughout our testing: virtually all of the SACD and DVD-A recordings sounded better than most CDs— sometimes much better. Had we not “degraded” the sound to CD quality and blind-tested for audible differences, we would have been tempted to ascribe this sonic superiority to the recording processes used to make them."


I just wanted to confirm what you understand by that quote Kimmo (just wondered if you were on the same page).

I understood it to mean...
There was no audible difference between these high-res recordings played back at high res, and played back at 16/44.1. The source material was better than most CDs, and we suspect that's due to the selection and mixing of the material - as we've confirmed it's not due to the use of high-res technology.

It wouldn't surprise me at all if this SACD and DVD-A recordings were mixed/mastered with higher dynamic content than regular CDs (and their over compression) because:
- The manufacturers wanted them to sound good and 'sell' the format
- Overcompression was unnecessary because high-res media will probably be played back on a high quality home audio system
 
The music originally created wasnt lossless, it was live.

A digital master would have been made for digital distribution and this would have been at 16bit/44100kHz or higher. This is what is given to the labels, Apple and everyone else as 'the tune'. This is what we should receive when we buy a tune, but Apple instead turns it into a compressed mp3 and sells it to us. I think this is fundamentally wrong in many ways.
 
A digital master would have been made for digital distribution and this would have been at 16bit/44100kHz or higher. This is what is given to the labels, Apple and everyone else as 'the tune'. This is what we should receive when we buy a tune, but Apple instead turns it into a compressed mp3 and sells it to us. I think this is fundamentally wrong in many ways.

Even though I've helped pull the conversation a bit away from the original post, I would have to agree that having Apple provide lossless 44.1k/16 bit versions of iTunes songs would be ideal.

Anything beyond that, though, is completely unnecessary.
 
I just wanted to confirm what you understand by that quote Kimmo (just wondered if you were on the same page).

I understood it to mean...
There was no audible difference between these high-res recordings played back at high res, and played back at 16/44.1. The source material was better than most CDs, and we suspect that's due to the selection and mixing of the material - as we've confirmed it's not due to the use of high-res technology.

Yes, I think we're on the same page about what they're trying to say. Later in the note they add:

"Plausible reasons for the remarkable sound quality of these recordings emerged in discussions with some of the engineers currently working on such projects. This portion of the business is a niche market in which the end users are preselected, both for their aural acuity and for their willingness to buy expensive equipment, set it up correctly, and listen carefully in a low-noise environment.

Partly because these recordings have not captured a large portion of the consumer market for music, engineers and producers are being given the freedom to produce recordings that sound as good as they can make them, without having to compress or equalize the signal to suit lesser systems and casual listening conditions. These re- cordings seem to have been made with great care and manifest affection, by engineers trying to please them- selves and their peers. They sound like it, label after label. High-resolution audio discs do not have the overwhelming majority of the program material crammed into the top 20 (or even 10) dB of the available dynamic range, as so many CDs today do.

Our test results indicate that all of these recordings could be released on conventional CDs with no audible difference. They would not, however, find such a reliable conduit to the homes of those with the systems and listening habits to appreciate them. The secret, for two-channel
recordings at least, seems to lie not in the high-bit recording but in the high-bit market."

Clearly, there is a lot of opinion and speculation in the note but it's interesting opinion and speculation. I think these guys did some good work, but a replication of the study (preferably in a university setting) and a report that goes into greater detail about the methodology used would be useful. With Professor Oohashi's "hypersonic" theory out there I personally won't be jumping on the "if the ear can't hear it, it doesn't matter" bandwagon based soley on the Meyers/Moran paper, but it sure is an interesting topic that's worth more time and discussion.
 
err...



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Actually, a live performance has all sorts of issues that are solved by a studio. Just the fact that the accoustics of a bar/stadium/arena/venue are not perfect distorts the analog waveform from the true original music.

Wikipedia is trustworthy. Trust me, I already tried changing little things like dates on it. Anyway, analog means the exact sound waves that were made. It is better quality, but digital has many other advantages.

We need EARDRUM DIGITAL (cannot tell the difference between it and analog).
 
Even though I've helped pull the conversation a bit away from the original post, I would have to agree that having Apple provide lossless 44.1k/16 bit versions of iTunes songs would be ideal.

That's a good idea D*I*S.

I think Apple would get a lot of positive feedback on that move ... at least until Neil Young starts demanding 32 bit floating point at 192 kHz. :)
 
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Even though I've helped pull the conversation a bit away from the original post, I would have to agree that having Apple provide lossless 44.1k/16 bit versions of iTunes songs would be ideal.

Anything beyond that, though, is completely unnecessary.

I don't particularly see the point in that for the reasons I've stated before. 256-320Kbps AAC is sufficient enough to be indistinguishable from 16-bit LPCM, predominantly because 16-bit LPCM is a pretty limited format. The only marginal benefit I can really see that would be worth the extra price they'd charge would be 24-bit LPCM for recordings that particularly demand a wider dynamic amplitude range, e.g. jazz, classical, instrumental soundtracks, older pop and rock recordings, etc.

Someone earlier had wondered about mastering standards for film... Dolby Digital, in particular, which is a 448Kbps 6-channel stream, has a series of requirements surrounding the metadata parameter settings (e.g. dialogue normalization, -3dB stereo down mix, DC noise filter, 20kHz lowpass anti-aliasing filter) but chief among these are the requirement that the soundtrack have an average A-weighted loudness, Leq(A), of -27dBFS. This is considerably below the average a-weighted loudness of sound recordings made in the last 20 to 30 years... which is why anything beyond 256Kbps AAC for most listening situations is completely superfluous.

What people do not realize is that AAC's data requirements are different for reconstruction of the same analogue wave because AAC is not just a compression algorithm but a perceptual coding algorithm that uses multiple techniques to eliminate redundant or imperceptible data.

Consider this: If you had an algorithm to break down how many times a given amplitude value appears in a stream, why would you need to represent that whole amplitude value every single time? Now multiply that times various types of coding algorithms all used together to truncate the data needed... as well as eliminate data that's not needed (e.g. anything outside the human range of hearing AND anything above the Nyquist limit which would cause unwanted frequency aliasing anyway... which is not always eliminated from a 16-bit LPCM CD master).

PCM doesn't have any method of intelligent truncation or perceptual coding... but why would you need every bit of data? If it takes 16 bits, let's say, to store a 28dB amplitude value at one sample, and another 16 bits to store a 27 at the next sample, why not eliminate 32 bits of data simply by storing the delta, and eliminate even further bits by using throttling (i.e. it only takes one bit to represent a delta of 1, so why store the other seven bits at that quantization interval when the result is EXACTLY the same?)

That's just one example... and there are many others. You can reduce the data quite a bit without losing any of the information, because the information in both cases is reconstructed from limited data... it's just that there are more mathematical tricks that have been devised since the advent of PCM to reconstruct the exact same information from fewer data.

I don't think there's enough of a benefit to justify full 16-bit uncompressed PCM, which would be quite enormous for the cost, at 1411 Kbps instead of 256 Kbps. And most purchasers of music would not pay the premium for that insignificant a difference (perceptibly). However, a substantially larger premium could be charged to the segment of listeners who tend to purchase music with a high dynamic range, if it were done in a 24-bit LPCM format that actually supported that dynamic range. That would justify the additional operating expense and data storage required.
 
I don't particularly see the point in that for the reasons I've stated before.... PCM doesn't have any method of intelligent truncation or perceptual coding... but why would you need every bit of data? If it takes 16 bits, let's say, to store a 28dB amplitude value at one sample, and another 16 bits to store a 27 at the next sample, why not eliminate 32 bits of data simply by storing the delta, and eliminate even further bits by using throttling (i.e. it only takes one bit to represent a delta of 1, so why store the other seven bits at that quantization interval when the result is EXACTLY the same?)

That's just one example... and there are many others. You can reduce the data quite a bit without losing any of the information, because the information in both cases is reconstructed from limited data... it's just that there are more mathematical tricks that have been devised since the advent of PCM to reconstruct the exact same information from fewer data.

I have no problem with more advanced data compression algorithms being used to generate the exact same final result. Maybe I should have phrased it "20hz - 22khz frequency response with 97db or better s/n indistinguishable from CD" instead of just "44.1k/16 bit." I wasn't thinking about better coding when I wrote that. I was focused more on the "overkill" threshold of the final audio result compared to unnecessarily higher sample rates and bit depths advocated by digital hi-fiers and the loony claims of the anti-digital/pro-vinyl crowd.

So my statement was definitely not "future-proofed" against better audio data compression schemes. Forgive me that. The overall point regarding required fidelity for human listeners stands, though, unless human evolution takes a radical new turn and increases the number and sensitivity of the cilia hairs in our cochlea such that we can hear more ultrasonic frequencies.
 
I was focused more on the "overkill" threshold of the final audio result compared to unnecessarily higher sample rates and bit depths advocated by digital hi-fiers and the loony claims of the anti-digital/pro-vinyl crowd.

Totally with you.

So my statement was definitely not "future-proofed" against better audio data compression schemes. Forgive me that.

Forgiven. I just was adding clarity for the fence sitters reading both our posts.

The overall point regarding required fidelity for human listeners stands, though, unless human evolution takes a radical new turn and increases the number and sensitivity of the cilia hairs in our cochlea such that we can hear more ultrasonic frequencies.

:D Exactly. I think what some of them are trying to argue is not so much perception of principal frequencies beyond the Nyquist limit as it is the idea that there may be harmonic resonance between base frequencies inside the A-weighted range and nth order harmonics that extend beyond it. But that's a moot point anyway because your ears wouldn't sum what they can't perceive. The research cited by some audiophiles earlier didn't really demonstrate a marked and consistent ability of obvious perception of such a phenomenon. The only area I think would be a very noticeable advantage of added bandwidth is in dynamic range OR multichannel surround.
 
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You know, I remember having a discussion recently with an ultrahigh frequency audio advocate who stated that people might be able to perceive ultrahigh audio content via other senses. I contended that ultrahigh frequencies can't be heard with human ears or "felt" very easily on one's skin via transmission through the air (unlike lows, which most certainly can) because the wavelengths are so short--BUT they could possibly be seen.

If a bat were to scream an inch or two from your eye, perhaps the surface of your cornea would undulate sympathetically in some weird fashion that you could see as wavy image distortion.

Not sure I want to test that myself, though! And I doubt visual distortion would be a good reason to include ultrahighs in audio codecs. If someone insisted on "seeing" sound, there are recreational pharmaceuticals that could assist them in such an experience...

:)
 
44.1k sample rate 16 bit depth PCM recordings = "eardrum digital"​

There you go.

Ah, but you are missing something. A channel for each instrument would be like real life, not just 2 channels. I should have mentioned that in my other post. Imagine if you could hook up your Mac or iPod to your 8-speaker home sound system and have each speaker be an instrument!

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A digital master would have been made for digital distribution and this would have been at 16bit/44100kHz or higher. This is what is given to the labels, Apple and everyone else as 'the tune'. This is what we should receive when we buy a tune, but Apple instead turns it into a compressed mp3 and sells it to us. I think this is fundamentally wrong in many ways.

It's lossy AAC now, not MP3, and higher quality. It used to be low quality MP3 with a stupid DRM on it. You are right, people need to quit compressing things lossily.

I got an analog AV (composite) to USB converter. Bad choice. The stupid software I needed to use it with compressed it to 320x240 MP4 or something so that I can "take it everywhere I go", and the quality was terrible. I later realized that my camcorder has the ability to convert the video input into digital video over FireWire! That was perfect.

Digital tape camcorders FTW.
 
Another interesting paper; this one from James Boyk. Professor Boyk ran Cal Tech's music lab. He is a concert pianist as well as a recording engineer of some note. http://www.cco.caltech.edu/~boyk/spectra/spectra.htm

Here is the discussion of his results:

"Given the existence of musical-instrument energy above 20 kilohertz, it is natural to ask whether the energy matters to human perception or music recording. The common view is that energy above 20 kHz does not matter, but AES preprint 3207 by Oohashi et al. claims that reproduced sound above 26 kHz "induces activation of alpha-EEG (electroencephalogram) rhythms that persist in the absence of high frequency stimulation, and can affect perception of sound quality." [4]
Oohashi and his colleagues recorded gamelan to a bandwidth of 60 kHz, and played back the recording to listeners through a speaker system with an extra tweeter for the range above 26 kHz. This tweeter was driven by its own amplifier, and the 26 kHz electronic crossover before the amplifier used steep filters. The experimenters found that the listeners' EEGs and their subjective ratings of the sound quality were affected by whether this "ultra-tweeter" was on or off, even though the listeners explicitly denied that the reproduced sound was affected by the ultra-tweeter, and also denied, when presented with the ultrasonics alone, that any sound at all was being played.
From the fact that changes in subjects' EEGs "persist in the absence of high frequency stimulation," Oohashi and his colleagues infer that in audio comparisons, a substantial silent period is required between successive samples to avoid the second evaluation's being corrupted by "hangover" of reaction to the first.
The preprint gives photos of EEG results for only three of sixteen subjects. I hope that more will be published.

In a paper published in Science, Lenhardt et al. report that "bone-conducted ultrasonic hearing has been found capable of supporting frequency discrimination and speech detection in normal, older hearing-impaired, and profoundly deaf human subjects." [5] They speculate that the saccule may be involved, this being "an otolithic organ that responds to acceleration and gravity and may be responsible for transduction of sound after destruction of the cochlea," and they further point out that the saccule has neural cross-connections with the cochlea. [6]

Even if we assume that air-conducted ultrasound does not affect direct perception of live sound, it might still affect us indirectly through interfering with the recording process. Every recording engineer knows that speech sibilants (Figure 10), jangling key rings (Figure 15), and muted trumpets (Figures 1 to 3) can expose problems in recording equipment. If the problems come from energy below 20 kHz, then the recording engineer simply needs better equipment. But if the problems prove to come from the energy beyond 20 kHz, then what's needed is either filtering, which is difficult to carry out without sonically harmful side effects; or wider bandwidth in the entire recording chain, including the storage medium; or a combination of the two.
On the other hand, if the assumption of the previous paragraph be wrong — if it is determined that sound components beyond 20 kHz do matter to human musical perception and pleasure — then for highest fidelity, the option of filtering would have to be rejected, and recording chains and storage media of wider bandwidth would be needed."
 
I manually shuffled them in different tabs of the browser until I could not identify which was which. I had to be careful not to hover or the popover filenames could give them away. I shuffled and covered when necessary, until I had no idea which was which. Then I listened to them back to back multiple times. Not good enough for publishing scientific results, but good enough for me.

I don't think it's good enough for anything. The ABX app is a free download, no excuse for not doing a listening comparison without it.

http://emptymusic.com/software/ABXer.html


analog means the exact sound waves that were made.

Absolutely not.

No microphones can capture the exact sound waves. And all analog recording media differ from the original sound waves in different ways and to different degrees. Limited dynamic range, limited (and not flat) frequency response, noise and in the case of records, clicks and pops, etc.

Digital has limitations as well, some are the same as analog, some different. But the notion that any recording medium is "exact", particularly to the sound in the room as opposed to the signal that came out of the microphones, is living in a fantasy.


A channel for each instrument would be like real life, not just 2 channels.

While that would be cool and would improve playback quality in many cases, still not "like real life" since each channel is subject to the same limitations. And it would be impractical since any recording can have any number of instruments. For things like classical recording it can be 100 or more players but still typically a relatively small number of microphones - in that situation generally individual mics would probably sound worse.
 
I actually find watching movies at 60 fps (and TV) is what is making me feel a bit queasy. It's just too natural looking and really messes with the suspension of disbelief.
But isn't that just something cultural that you'd get used to if you saw it more often?

We have HD now so things look crisper and more natural than ever before, so why are we then so stuck on a frame rate that makes everything stop looking natural again? Those HD demo videos that show natural history scenes at 60 fps in HD look amazing. Don't we want all our TV to look that good?

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Isn't Peter Jackson shooting The Hobbit at 48 fps? Theatres generally aren't set up for it yet but he's hoping there will be some installed before release. I believe Roger Ebert has seen material at higher framerate and is a huge advocate of it, he thinks it's much more of a visual boost than 3D.
That would be amazing if true. I'd love to see a movie in 48 fps, and see if people care about the difference it makes. It's great news if directors like Peter Jackson are pushing new technology in this way, hopefully it'll encourage others to follow suit in the future. :)

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If one were so inclined, one could build a Vinyl 2.0 standard with modern technology that would have staggeringly high specs. Whether it could reach 24/192 is unclear, but it could certainly blow past RedBook.
Whether that would be true or not, altering an analogue spec for a format like vinyl is a huge undertaking. The beauty of digital is that you can just add a few bits and a higher sampling rate and you're done. Admittedly the hardware would have to catch up if we went beyond 24/192.

Shooting at 24fps for movies makes as much sense as mastering recordings on 8-track - it's just old for old's sake. 1080p at reasonable viewing distance and high quality 60fps recording/playback gives a very visceral "3D" feeling - more so even, IMO, than the current silly-glasses approach.
And I would be much more inclined to see a movie shot at 48fps than a 3D movie shot at 24 :)
 
That would be amazing if true. I'd love to see a movie in 48 fps, and see if people care about the difference it makes. It's great news if directors like Peter Jackson are pushing new technology in this way, hopefully it'll encourage others to follow suit in the future. :)

<snip>

And I would be much more inclined to see a movie shot at 48fps than a 3D movie shot at 24 :)

I think shooting at 48 is basically a necessity for good 3D... it's no coincidence that it's effectively two 24fps captures interleaved. If he were shooting '2D', he'd probably have stuck with 24fps.
 
I think shooting at 48 is basically a necessity for good 3D... it's no coincidence that it's effectively two 24fps captures interleaved. If he were shooting '2D', he'd probably have stuck with 24fps.
Oh then that's a shame then. I don't see how you can call 24fps-3D 48fps, it's not running any more full picture frames every second.

EDIT: Hmmm that doesn't appear to be the case at all though... 48fps means 48fps for real:

http://www.firstshowing.net/2011/peter-jackson-talks-at-length-about-using-48fps-for-the-hobbit/

We've been watching HOBBIT tests and dailies at 48 fps now for several months, and we often sit through two hours worth of footage without getting any eye strain from the 3-D. It looks great, and we've actually become used to it now, to the point that other film experiences look a little primitive.

I rest my case! Higher frame rates are the future :D
 
Oh then that's a shame then. I don't see how you can call 24fps-3D 48fps, it's not running any more full picture frames every second.

EDIT: Hmmm that doesn't appear to be the case at all though... 48fps means 48fps for real:

http://www.firstshowing.net/2011/peter-jackson-talks-at-length-about-using-48fps-for-the-hobbit/

OK, that'll be interesting if it's 'proper' 48fps.

I'm not a great fan of 3D - but I've noticed that some of the regular 24fps 3D movies I've seen have been almost unwatchable... (like 'Pina'). Perhaps this will make 3D better.

I rest my case! Higher frame rates are the future :D

I'll wait 'till I've seen it to judge.
 
Some people like cars that are way more capable than they have a need for (other than the normal safety margin). In my case I feel the same way about computers and music. My Mac Pro is way more powerful and sturdy than I really need. It's the kind of computer I like.

Similarly I like music files that contain more data than my current equipment is capable of playing perfectly accurate. Maybe I can perceive the difference between 16-bit and 24-bit, maybe not. What I want is all of the music, that is as much of the music as is technically possible to have, in a file. If I have to downgrade it for iPhone use that's OK.

That's why I buy the highest quality files available at HDTracks. Maybe I'm overdoing it but as a consumer it's what I want to spend my money on.
 
I don't think it's good enough for anything. The ABX app is a free download, no excuse for not doing a listening comparison without it.

http://emptymusic.com/software/ABXer.html




Absolutely not.

No microphones can capture the exact sound waves. And all analog recording media differ from the original sound waves in different ways and to different degrees. Limited dynamic range, limited (and not flat) frequency response, noise and in the case of records, clicks and pops, etc.

Digital has limitations as well, some are the same as analog, some different. But the notion that any recording medium is "exact", particularly to the sound in the room as opposed to the signal that came out of the microphones, is living in a fantasy.




While that would be cool and would improve playback quality in many cases, still not "like real life" since each channel is subject to the same limitations. And it would be impractical since any recording can have any number of instruments. For things like classical recording it can be 100 or more players but still typically a relatively small number of microphones - in that situation generally individual mics would probably sound worse.

The mic limitations apply to digital also. I guess it's more accurate to say that "analog" means "whatever was passed into the mic and turned into electrical pulses". I know having a mic for each instrument wouldn't be practical for orchestras, but under the right conditions, you can do it. They should at least have 8 or 16 channels for a lifelike-ish recording.

You can't have perfect conditions, but you can come much closer than they do now. Lossy MP3 and AAC, only 2 channels? Lame.
 
They should at least have 8 or 16 channels for a lifelike-ish recording.
Where am I gonna put 16 speakers in my den? Do I arrange them in the front half of the room like a 4 piece string section and 4 piece wood ensemble? What if there's a piano? And a singer? What if there's a choir, or a drum set? Do you move them based on how you want the sounds reproduced? Seriously, where is anyone gonna put that many speakers? Talk about low WAF.

You can't have perfect conditions, but you can come much closer than they do now. Lossy MP3 and AAC, only 2 channels? Lame.

I'll agree with that. I use to think I wanted 24/192, but now I don't. A small part of me still wants 24/96, but I'd be happy being able to buy well mastered ALAC albums at 16/44.1.
 
Ah, but you are missing something. A channel for each instrument would be like real life, not just 2 channels. I should have mentioned that in my other post. Imagine if you could hook up your Mac or iPod to your 8-speaker home sound system and have each speaker be an instrument!

The reason we have two channel stereo audio is because we have two ears. The illusion of a three dimensional soundstage in which one can clearly identify the relative positions of many different sound sources is created by our ears and our brains.

We don't need eight separate speakers to emulate the sound of an eight-piece Dixieland Jazz band. Just two.

Surround is to a great extent a gimmick. It is useful in large movie theaters, where 300 people have to fit in the stereo monitoring "sweet spot", and even in that application it is most effectively used just for sound effects and ambience. I like the ".1" of "5.1" surround, as omnidirectional low end can help make a recording sound truer to the real thing, and for movies a center channel dedicated to speech is nice, but for music listening a two channel stereo mix fits the bill. Any respectable system can crossover the FR speakers and direct lows to a sub, so even a 2.1 mix is not really necessary.

While it might be retro fun to hear DSOTM in 5.1 surround (just a modern version of 70's "quadrophonic"), I'm not sure I would want to bother with it for many other albums.
 
Ah, but you are missing something. A channel for each instrument would be like real life, not just 2 channels. I should have mentioned that in my other post. Imagine if you could hook up your Mac or iPod to your 8-speaker home sound system and have each speaker be an instrument!

This would mean that instruments would no longer be able to switch sides or anything anymore though. Depending on the kind of music you listen to you would have to rearrange the speakers in your room prior to listening.

Also the sound would sum up in the worst place possible: your room. Unless you have a perfectly sound-treated room at home the results would be disastrous.
 
The mic limitations apply to digital also.

Which is exactly what I said.

I guess it's more accurate to say that "analog" means "whatever was passed into the mic and turned into electrical pulses".

Nope, wrong again.

Going from acoustic sound waves to a mic changes the sound, it can't be captured perfectly.

Going from the electrical signal to any recording medium, digital or analog, changes the sound to some degree.

You just need to let go of the notion that "analog" means making an exact copy, it has no basis in reality.

The reason we have two channel stereo audio is because we have two ears.

But even with two ears we have some perception of in front and behind us. Not to mention that people move their heads around when they listen which aids in the front/back imaging.

Also, it has been shown that more speakers generally sound better, all else being equal - splitting a mix over more channels fed to more of the same speakers are perceived as having higher quality sound. Not that it would make sense to go to a speaker per instrument, but the 5.1 setups many people have would sound better with music mixed to that format. Too bad a surround music format hasn't really caught on, but I can see why with all the mobile listening people do - the best shot for it would probably be a download format that included stereo and surround versions of the same files.
 
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