Newman Posted January 9, 2016 Posted January 9, 2016 Simply cannot see how using the highest bit rate is not a good thing, particularly as digital storage is becoming so affordable. Why we are not buying hi res digital recordings on USB SSD, is beyond me. I see the CD and its transport's days coming to an end. High res download sites are charging a premium for 24/96 and up, typically double the price for standard def. That means you can only get to buy half as much music for your budget. If the audible difference is either nothing or a pooftenth of bugger-all, then having your library cut in half is IMHO an enormous penalty for going high-res. Almost as big as the penalty for going (new) vinyl. So, the high res movement has got a lot to answer for, so their claims had better be more than academic. 4
Martykt Posted January 9, 2016 Posted January 9, 2016 They're telling you that high sampling rates are giving you "more information" .... is that really the case? My understanding is yes more samples = more information. More values in time are recorded."For functions that vary with time, let s(t) be a continuous function (or "signal") to be sampled, and let sampling be performed by measuring the value of the continuous function every T seconds, which is called the sampling interval.[1] Then the sampled function is given by the sequence: s(nT), for integer values of n. The sampling frequency or sampling rate, fs, is the average number of samples obtained in one second (samples per second), thus fs = 1/T."
Volunteer sir sanders zingmore Posted January 9, 2016 Volunteer Posted January 9, 2016 If I have my understanding of how digital sampling works correct (not going to pretend to be anything more than a layman) a higher sample rate will be able to create a more precise sound wave. The more samples T there are the more precise X will be as it is recorded at that point in time. No. That is a very common misconception but it is wrong. The analog waveform between 20Hz - 20kHz (and a bit) can be perfectly reconstructed at the standard redbook sampling rate
firedog Posted January 9, 2016 Posted January 9, 2016 If I have my understanding of how digital sampling works correct (not going to pretend to be anything more than a layman) a higher sample rate will be able to create a more precise sound wave. The more samples T there are the more precise X will be as it is recorded at that point in time. Nope. Those "stepped" graphs we always see are not an accurate reflection of what's happening. It's just a graphic used to explain what's going on, that unfortunately is misleading. Nyquist theory proves mathematically that a sampling rate of 2X the highest frequency can perfectly produce the sound wave between any two points - in real life the there are no "steps" in the resulting wave. So higher res isn't more accurate in the sense you are describing. (the theory also assumes infinite perfect filtering, etc, but that is irrelevant to your point). The rationale for hi-res for me is it enables the use of much better filters which get rid of aliasing and other digital artifacts that occur in real life. So hi-res enables us to use better filtering, which gets us much closer to the "perfect conditions" assumed by the Nyquist theory. Thus it sounds better, more natural to our ears. It may also do what the Meridian people say, which is better allow us to hear transients. I have no idea if that is correct. 1
Guest rmpfyf Posted January 9, 2016 Posted January 9, 2016 Wow. Some major misconceptions in the video. It's true that there are advantages in recording/mastering at higher frequencies but there's no reason it's needed for playback assuming your analogue signal reconstruction. A higher-frequency recording will sound better if filtering is improperly applied to effectively LP anything north of 20kHz. Just because you can open a discrete time history in Audacity, or because digital recording is a discretised process, does not mean that this is what a DAC recreates nor what goes through your speakers, through air and to your ears. That's BS. From your DAC forwards it's about making a continuous signal out of a discretised time history. Anything can effectively LP-filter what happens before, in and beyond this process - from mechanical or electrical interference, electromechanical response, amplifier slew rates, whatever. If you could 'locate' two discrete sounds at a time delay of triple-digit-kHz... you would have frequency response exceeding 20kHz. You don't. Whether content exists above 20kHz in a recording or not is a moot point if you can't hear it, and claiming audible differences for a higher-frequency version of a same recording where spectral content below 20kHz is identical simply means whatever your system is doing to reproduce 20kHz isn't enough, because it's adding something you can actually hear within the very content it's supposed to reproduce. The only change in spectral content you can hear is under 20kHz. If you can pick a difference with high def, MQA, DSD, etc... and if you want to improve your system... address it's ability to do the first 20kHz faithfully. There are some difficult things in Redbook - it's less amenable to low-cost implementations, digital volume control - but done right there is no need for anything more. Higher frequency playback makes it easier - which in itself can be better - but not for the reasons put forwards in this video. This comment is 100%: Its the down sampling which is the problem. This happens not just in how music is stored but effectively - to many ends - in how it's recreated from digital and played back at every stage. If MQA proponents want to talk about improved group delay in playback and more flexibility in filter design from higher sampling frequency, fine, but the stuff skirting on the fringes of 'you can hear more than 20kHz, maybe, if we call it something else' is faff. <hides back under a table>
Martykt Posted January 9, 2016 Posted January 9, 2016 No. That is a very common misconception but it is wrong. The analog waveform between 20Hz - 20kHz (and a bit) can be perfectly reconstructed at the standard redbook sampling rate Remember we are talking about the reconstruction of a more complex music audio wave not a single frequency wave.It has been stated in the article that yes 44.1 khz does have the frequency wave limits 20hz - 20khz. That is not in question. That said my understanding of how digital sampling works as shown I am basing off the information provided in Wikipedia (easier to understand for the layman) and the link that @@davewantsmoore has provided. That said I am happy for you to go more in-depth about what the common misconceptions you believe there are and where you feel I may be mistaken. I'm always for rather than against accurate information and technical understanding.
Guest Posted January 9, 2016 Posted January 9, 2016 High res download sites are charging a premium for 24/96 and up, typically double the price for standard def. That means you can only get to buy half as much music for your budget. If the audible difference is either nothing or a pooftenth of bugger-all, then having your library cut in half is IMHO an enormous penalty for going high-res. Almost as big as the penalty for going (new) vinyl. So, the high res movement has got a lot to answer for, so their claims had better be more than academic. Is ok, I bow out of this discussion and respect everyone for their input. Understand quite well there is a poofteenth of bugger all as an audible difference between higher resolution and red book, but there is a difference if the master was made to be of superior sound quality. If low resolution had not been invented I would not own a big pile of poor sounding MP3 files, which are slowly being replaced. After listening to a nice TT with a healthy cart. the odds of running an MP3 after are slim as the audible difference is astonishing. CDs are Ok, but sometimes a recording on vinyl of the same artist, same track sounds better on the same high end / system speaker set. Would be happy if I listened to a blu tooth bogan blaster, because the subtle difference would not be audible. Digital has the potential to be on paper as good as analogue, then why not ? Agree that the Hi Res movement has a lot to answer for, but they are probably making more money selling vinyl and don't care.
Nada Posted January 9, 2016 Posted January 9, 2016 Remember we are talking about the reconstruction of a more complex music audio wave not a single frequency wave... It makes intuitive sense to me too that music being such complex modulated variable might need higher sampling rates to recreate the actual wave form precisely then single frequency encoding would allow. But its just not true. 44.1k does it as precisely as 196k assuming all sounds are sine waves and its pretty hard for longitudinal air waves not to be sine waves. Thats the beauty of nature. Its modelled so astonishingly well by math. Its like there is logic at the core of everything. I guess God is pure math? 5
aechmea Posted January 9, 2016 Posted January 9, 2016 I suspect that some value may come of this if the music has more attention paid to it in the recording, mixing and mastering processes. If its the same old heavy handed compression then there is no real point. 3
Volunteer sir sanders zingmore Posted January 9, 2016 Volunteer Posted January 9, 2016 I suspect that some value may come of this if the music has more attention paid to it in the recording, mixing and mastering processes. If its the same old heavy handed compression then there is no real point. If more attention is paid to recording/mixing/mastering then that is what will cause it to sound better, not the fact that it's "Hi Rez" 5
LHC Posted January 9, 2016 Posted January 9, 2016 Remember we are talking about the reconstruction of a more complex music audio wave not a single frequency wave. It has been stated in the article that yes 44.1 khz does have the frequency wave limits 20hz - 20khz. That is not in question. That said my understanding of how digital sampling works as shown I am basing off the information provided in Wikipedia (easier to understand for the layman) and the link that @@davewantsmoore has provided. That said I am happy for you to go more in-depth about what the common misconceptions you believe there are and where you feel I may be mistaken. I'm always for rather than against accurate information and technical understanding. I think the problem with the communication here is that the posters are not first establishing where they might actually agree. They are talking about things they disagree with, so there is no way to build up a common understanding of what are the issues here. I have seen this communication problem many times on SNA forum. Ultimately it just lead to cross purposes shouting and no one is more the wiser. These are what I am confident are facts: Few if any human could hear frequency signals above 21kHz, by that I mean no one can say "hey, I can hear that 37kHz sound". Mathematically a 44kHz sampling rate can reproduce all the 'frequency information' below 21kHz IF the recording engineers use a limiting bandwidth of not recording above 21kHz. This choice of bandwidth is completely arbitrary (i.e. Alien from outer space may chose a different bandwidth), but the reason is based on the first dot point above. If we want a higher bandwidth, say around 48kHz of frequency information, then the mathematics tell us we should use a sampling rate that is double, i,e, we have to sample at 96kHz. No one can question this. The higher bandwidth desired, the higher sampling rate is required (using the doubling rule each time). There is no doubt that the higher bandwidths (and hence higher sampling rate) will give you more frequency content. That is a mathematical fact, not a misconception. The extra information will you bring closer to the original music that was sampled; the resultant waveform more resembles the original waveform than the CD sampling of 44kHz. (this of course assumes the original music contains frequency information above the audible 21kHz). More is better in an academic sense. More specifically when we look at how the sound wave evolves in time, i.e. looking at the time domain, the extra frequency information helps one build up much sharper waveforms. Waveforms from the CD sampling of 44kHz has a limitation to their sharpness due their lower bandwidth. Take a look at that diagram of peaks posted by Nada. Sharper curves means the rise time of the signal (or musical note) is much more quicker, and the notes are snappier and better defined. This is a fact, not a misconception, but in an academic sense. I believe these points are agreeable to everyone here including those well versed in digital sampling theory. Now where I think the disagreement starts is the following: Academically we are suppose to gain extra benefit from higher sampling rate and larger bandwidth. But what happens in practice? In particular give the first dot point that we can't hear frequencies above 21kHz? Some people will argue that it is a given that the audible limit automatically implies we can't ever reap the benefit of the extra information in the frequency domain and time domain. What we hear from higher sampled music is exactly the same as what we hear from CD Others may propose that we can't reap the benefit of extra frequency information in the frequency domain (we still can't say 'I can hear that 37kHz sound'), BUT we can reap the benefit of higher rise time in the time domain and sharper notes. This is the basis of the MQA proposal. They claim our sensitivities to timing is far above frequencies. I hope this helps, but it may causes more confusion and angsts. 5
betty boop Posted January 9, 2016 Posted January 9, 2016 you tube videos aside....and all the philosophical arguments(notice I didnt say scientific). if there is genuine advantage to be had...I think most people tend to go with it. and am talking a serious actual advantage...not just up sampling stuff to 192khz ! I too can rip cds and vinyl to 192khz. why would I bother ? is there genuinely a lot of true 192khz stuff out there for this to be even a discussion worth having ? 1
LHC Posted January 9, 2016 Posted January 9, 2016 is there genuinely a lot of true 192khz stuff out there for this to be even a discussion worth having ? Ain't all movies on DVD and Blu-ray contain 192kH tracks? There is a massive amount of material out there.
betty boop Posted January 9, 2016 Posted January 9, 2016 Ain't all movies on DVD and Blu-ray contain 192kH tracks? There is a massive amount of material out there. DVD ? no way ! as far as blu-ray goes ...in the entire collection known to man...there are a grand total of 6 titles with 192khz audio...see below, Casablanca Warner 2009-09-15 8.8 $1.7M VC-1 BD50 DD 1.0 192kHz $28.99 La Voie Triomphale Other 2013-01-29 DTS-MA 5.1 24bit 192kHz Yes $34.99 $33.79 Neil Young Archives, Vol. 1: 1963-1972 Warner 2009-06-02 AVC BD25 LPCM 2.0 24bit 192kHz $349.99 Percy Grainger/Kristiansand Symfoniorkester: Grieg - Piano Concerto Other 2009-10-27 DTS-MA 5.1 24bit 192kHz Yes $34.99 Schubert: Winterreise Other 2013-07-09 LPCM 5.1 24bit 192kHz $29.98 $29.98
aechmea Posted January 9, 2016 Posted January 9, 2016 (edited) If more attention is paid to recording/mixing/mastering then that is what will cause it to sound better, not the fact that it's "Hi Rez" Yep, that's what I thought I said. Well, that was my intention anyway. Clumsy wording perhaps. My Reference Recordings (HDCD and HRX) are very nice indeed because of the work put in by Prof Johnson rather than them being 20 bit and 24/192 respectively. (Edit: Doesn't matter anyway 'cos the music is not to my taste.) Edited January 9, 2016 by aechmea
Jake Posted January 9, 2016 Posted January 9, 2016 DVD ? no way ! as far as blu-ray goes ...in the entire collection known to man...there are a grand total of 6 titles with 192khz audio...see below, Casablanca Warner 2009-09-15 8.8 $1.7M VC-1 BD50 DD 1.0 192kHz $28.99 La Voie Triomphale Other 2013-01-29 DTS-MA 5.1 24bit 192kHz Yes $34.99 $33.79 Neil Young Archives, Vol. 1: 1963-1972 Warner 2009-06-02 AVC BD25 LPCM 2.0 24bit 192kHz $349.99 Percy Grainger/Kristiansand Symfoniorkester: Grieg - Piano Concerto Other 2009-10-27 DTS-MA 5.1 24bit 192kHz Yes $34.99 Schubert: Winterreise Other 2013-07-09 LPCM 5.1 24bit 192kHz $29.98 $29.98 Only 6? I know I've got some and they are not on your list.Anyway, interesting thread, will read it all tonight to help me sleep.
LHC Posted January 9, 2016 Posted January 9, 2016 DVD ? no way ! as far as blu-ray goes ...in the entire collection known to man...there are a grand total of 6 titles with 192khz audio...see below, Fair enough, I stand completely corrected by my oversight. In any case those blu-ray will still contain tracks of higher than standard CD rate, something like 24 bit/96 kHz. So this is still relevant to the discussion of this thread.
Martykt Posted January 9, 2016 Posted January 9, 2016 These are what I am confident are facts: [*]There is no doubt that the higher bandwidths (and hence higher sampling rate) will give you more frequency content. That is a mathematical fact, not a misconception. The extra information will you bring closer to the original music that was sampled; the resultant waveform more resembles the original waveform than the CD sampling of 44kHz. (this of course assumes the original music contains frequency information above the audible 21kHz). More is better in an academic sense. [*]More specifically when we look at how the sound wave evolves in time, i.e. looking at the time domain, the extra frequency information helps one build up much sharper waveforms. Waveforms from the CD sampling of 44kHz has a limitation to their sharpness due their lower bandwidth. Take a look at that diagram of peaks posted by Nada. Sharper curves means the rise time of the signal (or musical note) is much more quicker, and the notes are snappier and better defined. This is a fact, not a misconception, but in an academic sense. Yes, those two points are pretty much what I was trying to get at. (You've written it much better though )
betty boop Posted January 9, 2016 Posted January 9, 2016 Fair enough, I stand completely corrected by my oversight. In any case those blu-ray will still contain tracks of higher than standard CD rate, something like 24 bit/96 kHz. So this is still relevant to the discussion of this thread. for 96/24 that list in movies grows to 22 ! thats out of a total of 4500+ blu-rays released about 0.34% according to the blu-ray stats site .... puts into context 1
Guest rmpfyf Posted January 9, 2016 Posted January 9, 2016 (edited) More specifically when we look at how the sound wave evolves in time, i.e. looking at the time domain, the extra frequency information helps one build up much sharper waveforms. Waveforms from the CD sampling of 44kHz has a limitation to their sharpness due their lower bandwidth. Take a look at that diagram of peaks posted by Nada. Sharper curves means the rise time of the signal (or musical note) is much more quicker, and the notes are snappier and better defined. This is a fact, not a misconception, but in an academic sense. This suggests that a linearised signal (spectral) at higher frequency can be used to better approximate a waveform (time domain) where that waveform is especially sharp. This argument is academic, and only practically valid in recreating waveforms not associated with music (square waves etc), the replication of which is are more practically constrained by many other factors (output transformer design, electromechanical systems etc). In short, if you could increase Fourier density to a point where a near-as-dammit infinitely sharp waveform could be recreated in a discrete time history, you'd neither the means to hear it nor an instrument with which to generate it in an audible sense (or one to record it). Academically we are suppose to gain extra benefit from higher sampling rate and larger bandwidth. But what happens in practice? In particular give the first dot point that we can't hear frequencies above 21kHz? Some people will argue that it is a given that the audible limit automatically implies we can't ever reap the benefit of the extra information in the frequency domain and time domain. What we hear from higher sampled music is exactly the same as what we hear from CD Others may propose that we can't reap the benefit of extra frequency information in the frequency domain (we still can't say 'I can hear that 37kHz sound'), BUT we can reap the benefit of higher rise time in the time domain and sharper notes. This is the basis of the MQA proposal. They claim our sensitivities to timing is far above frequencies. An interesting middle ground, though a few caveats. We can't really, honestly, directly compare hires and, say, Redbook. We can't test that we might hear "exactly the same as what we hear from CD". If a file comes from the same master, the process to simply create a Redbook and, say, 192kHz version introduces differences throughout the entire spectral content. Why? Filtering isn't just about amplitude response, phase response is a major component and needs to be managed (hence the earlier comment on group delay). The digital filter employed to make Redbook possible is a major achievement and that was the late 70's/early 80's. Any filter design is an inherent compromise. Filters exist in mastering, in downsampling, and effectively throughout your entire system from source material to your ear drums. Some filers we take fore granted that cause mega angst to SQ - sink resampling, volume control. Most of us have zero idea of what's inherent filtering on our DACs (but it's there, it's complex, and DAC designers lose hair over how best to implement it - really). Very few of us have played a controlled source and measured the audible frequency response where we sit and listen to music, and for those of us that have, even fewer have understood phase response. This is important to understand because hires audio allows a shedload more flexibility in filter design. The ideal filter (for mastering material for playback) allows 0-20kHz to go through at unity (unchanged) amplitude and phase. That's impossible, and there's around ~4khz of room in Redbook to effect the filter (without a digital filter it'd not be possible at all). Redbook done right gets very, very close on filter quality. Close enough to be broadly inaudible - that's by design. Note I wrote 'done right'. It's easy to stuff up. Consider that the first Redbook DACs are regarded today as reference equipment. Start some hardcore cost reduction and you don't get close. Run on ~30 years of it and Redbook is not easy to get right. Take computer audio - most people don't actually realise (e.g. hello Linux users) that the upper response limit of the audio manager (Pulseaudio) in default form, when playing Redbook, is around 17khz. As in there is nothing reproduced after 17khz and there's a rolloff curve before it. Run some hires through that though (even some 48kHz) and yes, on account of how the same filter works over a different sampling frequency, you will hear more content. Run some 192kHz through it and it's downright sparkly. It's not that the content wasn't there to start with though. Digital distortions are more present than ever in the transports we use relative to the things we used to listen to. A good oscillator in a good CD transport has a very simple job to do - play CDs! The crystal is literally cut at a frequency that's a multiple of 44.1khz. But a PC? It's a noisy-as-hell environment and the crystals are cut to be 'good enough to run a PC'. You'll get more jitter out of a PC at 44.1kHz every time. There's less, proportionately, to be had at higher frequencies (assuming you can sustain data transfer to your DAC) simply because the next sample arrives sooner. Yes, you can remedy these issues... but how many of us actually run a reclocker? Everything a**es up frequency response from your cables to small vibrations be it in analogue or digital domains. Hires allows lazier filters but it doesn't fix underlying problems in the first 20kHz, which is all that matters. Honest. It can allow lazier filters in both mastering and reproduction, which can make for systems either way that are easier to implement. But so long as the content is there and hasn't been attenuated too much it can be corrected to the point that the right stuff gets to your ears. For the rest of it, spend efforts ensuring the content gets there. Newer audiophile DAC chips exist to make playback better, to more faithfully reconstruct and to better filter. The means for either are not higher sampling frequency. The best system I've ever heard (it's an hour north of Sydney)... runs Redbook. If the rest of your system is right, it is all that's needed. Golden rule: if you can play Redbook really well, you'll play anything really well, and it won't sound too different. Really. The rest just gives more flexibility, but nothing will sound better than an ultimate Redbook system... because there is nothing else to hear. The 'ultimate' Redbook system is as much a unicorn as a 'best' DSD or MQA system. The same basic rules apply - do what you can to get every last bit to your DAC, do what you can to correct it the rest of the way to your ears as it's makers intended. Whatever the sampling frequency. Edited January 9, 2016 by rmpfyf
Newman Posted January 9, 2016 Posted January 9, 2016 (edited) Is ok, I bow out of this discussion and respect everyone for their input. Understand quite well there is a poofteenth of bugger all as an audible difference between higher resolution and red book, but there is a difference if the master was made to be of superior sound quality. If low resolution had not been invented I would not own a big pile of poor sounding MP3 files, which are slowly being replaced. After listening to a nice TT with a healthy cart. the odds of running an MP3 after are slim as the audible difference is astonishing. CDs are Ok, but sometimes a recording on vinyl of the same artist, same track sounds better on the same high end / system speaker set. Would be happy if I listened to a blu tooth bogan blaster, because the subtle difference would not be audible. Digital has the potential to be on paper as good as analogue, then why not ? Agree that the Hi Res movement has a lot to answer for, but they are probably making more money selling vinyl and don't care. Even CD is better than any analog recording/playback medium, no question. To set the bar 'as good as analog', digital tech would have to go way backwards. It's the wrong goal, entirely. Leaving analog in the dust, the goal for future audio distribution is a format that resolves uncompressed [edit: I meant to say lossless, not uncompressed] 20/48 in an unlimited/unspecified number of channels, a kind of sonic 'raw image' file containing 360Hx360V surround information, that the playback system can reconstruct into as many channels as the user can accomodate, beyond Atmos, to whatever the future may bring, and in a sense be 'future-proof' or 'upgrade-ready', and to have this 'raw' format be of a manageable size for download/streaming. Pardon my dream. I don't even know if anyone is working towards that. But, 'as good as analog' plays no part in it, for sure. Edited January 9, 2016 by Newman 1
huxmut Posted January 9, 2016 Posted January 9, 2016 Have a watch of Monty This guy makes it easy to understand sampling 5
Nada Posted January 9, 2016 Posted January 9, 2016 (edited) ........More specifically when we look at how the sound wave evolves in time, i.e. looking at the time domain, the extra frequency information helps one build up much sharper waveforms. Waveforms from the CD sampling of 44kHz has a limitation to their sharpness due their lower bandwidth. Take a look at that diagram of peaks posted by Nada. Sharper curves means the rise time of the signal (or musical note) is much more quicker, and the notes are snappier and better defined. This is a fact, not a misconception, but in an academic sense. Oh dear. Im afraid that notion above is completely mistaken. Indeed its the core myth that certain audio brands try and fool us with. So to follow it with I believe these points are agreeable to everyone here including those well versed in digital sampling theory. is mistaken. Look, just because a diagram looks good and seems to make sense is no reason to get fooled. If you want to understand this issue its essential to understand that diagram I posted is totally misleading unless you are wanting to make sounds for a bat. I posted it as an educational prop. Impulses below 22kHz that we can hear ie musical notes are no sharper or better timed above 44.1kHz sampling. So 96, 192 or DSD Mhz sampling cannot make any difference to what you can hear. Why? Because the time domain impulse is set by the frequency in a simple inverse relationship. To repeat then the diagram below while factually true, is an example of the deceit in marketing. In terms of music its a lie. Edited January 9, 2016 by Nada 1
Tommy107 Posted January 9, 2016 Posted January 9, 2016 Well it should be buy the best gears you could afford n then just sit down n enjoy the music of whatever you have . Who care which sound better or what will stop , it all about how much n what you want to listen in any recording . It heathy to have a debate but again IMO it the recording that mater not high res , 44k or 192k . If it good it will sound good at any rate ( not MP3 ) [emoji6]
Guest rmpfyf Posted January 9, 2016 Posted January 9, 2016 Have a watch of Monty This guy makes it easy to understand sampling Great video (I used to own that CRO! Want to find another one now). His bit from 21 minutes onwards debunks part of the impulse response argument, though remember, 'sharpness' is as good as the master recording. I'd suggest anyone wanting more from the MQA tech - as in reading the line above and inferring that MQA is 'sharper' - might want to go research basics on transformer behaviours (ringing etc at high frequencies in particular) or, more generally, amplifier slew rates, electromechanical (e.g. loudspeaker) response, room transmission and frequency response, etc. In short: the odds of being able to reproduce a square wave faithfully - even at infinite sample rates - zero. The odds of wanting to listen to one as such are probably less! I own a lot of music... no square wave content!
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