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DSD [A Reprise]...

213.137.8.6
Posted on September 11, 2014 at 06:47:22
Storris
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Posts: 32
Location: London
Joined: September 8, 2014
 I'm just gathering my thoughts about DSD. I thought I'd gather them here in order to gain corrections/pointers from people who actually know what they're talking about.

I'm not a young pup by any means (35), but this time last month I didn't know what DSD was. I knew about flac, alac and mp3, wav and aiff, but had never heard of DSD. I knew high fidelity music existed, I thought I was listening to it! Then I had a run in with Windows Media Player & took to scouring message boards to find a replacement. On the way I found DSD and I think most importantly, I started to question what I was listening to. I'm now upgrading as many of my music files as I can find high quality sources for. I'll still be listening through my PC or Smartphone connected to Bose headphones though...

Everything I've read to-date, approximately two decades' worth of data-sheets, Msc & DIY projects and forum posts, suggests that the conversion, recording, transport and playback of audio via DSD is, or at least should be, a simpler and cheaper solution to A/D/A than via PCM.

On that assumption (my background is in mechanical rather than electrical/electronic/audio engineering) I have based the following...

It seems that the reason DSD is a niche format is a combination of factors including the inherent size of DSD files, Sony/Philips early crushing grip of the format, and the general lack of knowledge in the public about DSD.

With continuous growth in StorageCapacity/in²/£ allowing multi-TB Hard Drives to become the norm, and with 10's of MB/s of internet bandwidth available to many, it is possible for even the largest digital music libraries to be accumulated easily and stored entirely on the home PC, the size of DSD files is now almost an irrelevance.

The proprietary grip on the format no longer exists. Innovation has seen new solutions to DSD popping up, from PS Audio's hardware to Merging Technology's software. There are plenty of fish in the sea, even if they are all swimming around in the same small area! The sparse DSD music landscape (thanks to Sony's heavy boot?) is a concern, but by all accounts, DSD done well has the potential to make other digital audio formats sound better (does this also hold for MP3?), increasing the potential immediate market for DSD capability beyond those few who currently own DSD files.


And this brings me I think to the biggest PITA...


As I said, I based these thoughts on the premise that DSD was easier, simpler and cheaper to do than PCM. But the material evidence, the DSD DAC market, says otherwise.

Inside my PC, a mid-sized tower with discrete graphics card, I currently have enough spare, empty and usable space (which also happens to accommodate 2 available PCI slots) to place a small retail DSD-DAC, case and all. I believe that with a little real-estate design and component-placement management, PS Audio's Directstream hardware would also fit nicely. My point? That as a minimum a discrete DSD capable PC sound card is in the the realms of possibility, physically at least.

I presume that if it was possible, a software-only solution would already have been produced. Still, I can't help thinking that the simplicity of the required circuits, the power of computers and the genius of programmers, should be able to piece together a high quality application. Am I missing something fundamental here? Is it possible that people can create and programme algorithms capable of predicting a system as chaotic as the weather, but are unable to simulate a suitable low-pass filter who's parameters are well defined and well understood?

This is a massive assumption, but I suspect that given the option people would rather listen to hi-fidelity music, than watch high-definition videos of cats, perhaps. But they aren't going to be able to purchase products that they don't know about or that they have no way to access. If the market for hi-fidelity sound is to mature beyond the confines of the audiophile's 'listening room', there will have to be at least some attempt to market it by the industry. They will also need something to sell and £000's boxes aren't going to cut it in volume.

DSD is in a complex and expensive/time consuming niche, when it could, should be the standard.

 

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RE: DSD [A Reprise]..., posted on September 11, 2014 at 07:11:02
Tony Lauck
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Posts: 13629
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 Just get a USB capable DAC and connect that to your computer. Then load appropriate software and you will be able to play DSD files.

There are many posts on this forum regarding DSD DACs and suitable software.




Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar

 

RE: DSD [A Reprise]..., posted on September 11, 2014 at 08:39:16
AbeCollins
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That as a minimum a discrete DSD capable PC sound card is in the the realms of possibility, physically at least.

But why put such a device inside a harsh noisy environment? Most choose to use outboard DACs. Some DSD capable DACs like this one cost under $200 USD. There are several under $1000 USD.

http://ifi-audio.com/portfolio-view/nano-idsd/

You can get DSD player software for smartphoness including the Onkyo HF Player:
http://www.intl.onkyo.com/downloads/applications/hf_player.html
But the files are still huge so IMHO, they don't belong on a smartphone.

If the market for hi-fidelity sound is to mature beyond the confines of the audiophile's 'listening room', there will have to be at least some attempt to market it by the industry. They will also need something to sell and £000's boxes aren't going to cut it in volume.

Sony invented DSD and they are marketing it along with:

http://discover.store.sony.com/High-Resolution-Audio/


 

RE: DSD [A Reprise]..., posted on September 11, 2014 at 11:29:09
Tony Lauck
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 "Sony invented DSD"

Beg to differ. Sony and Philips invented SACD, the optical disk and associated encoding and DRM technology. They invented some editing techniques. However, DSD is just one bit PCM at a high sample rate and the methods of generating it (sigma delta modulators) were previously known. What they did do was standardize a bit rate, recording levels, and placed requirements on in band and out of band noise for acceptable implementations.

Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar

 

This seems easy, posted on September 11, 2014 at 11:45:59
G Squared
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 and cheap if you want to try.
Gsquared

 

RE: DSD [A Reprise]..., posted on September 11, 2014 at 15:23:01
AbeCollins
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 According to the link below Sony invented DSD.

 

RE: DSD [A Reprise]..., posted on September 11, 2014 at 19:01:58
Storris
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Posts: 32
Location: London
Joined: September 8, 2014
 Thanks for the link Abe...

...but it sort of proves my point. I'm practically middle aged and have never seen those stores before. On the market sure, but not what you'd call marketing.

Why not put such a device inside a harsh noisy environment?

The inside of a PC is full of sensitive electronic equipment carrying functionally vital digital signals, and it all works perfectly. I don't know if PDM is any more susceptible to noise than the other digital signals in a PC, or if the inside of a DAC/amp/speaker cabinet and their many cables and connections are any less harsh, but one benefit of PDM seems to be that unwanted noise is inherently simple to control.

 

RE: DSD [A Reprise]..., posted on September 11, 2014 at 19:53:52
Tony Lauck
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 Your link was rehashing Sony marketing hype. The underlying technology (Sigma delta modulation) was invented in the early 1960's, if not before.

If you read the Wikipedia Article you will get a reference to an early IRE article.




Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar

 

RE: DSD [A Reprise]..., posted on September 11, 2014 at 22:18:53
AbeCollins
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"Why not put such a device inside a harsh noisy environment?"

IMHO, because the analog sections of the DAC should not be in that harsh noisy environment.



 

RE: DSD [A Reprise]..., posted on September 11, 2014 at 22:27:41
AbeCollins
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 Sure Sony invented "DSD". They are credited with developing and bringing practical application of sigma-delta modulation ADC to the audio entertainment industry, and they called it DSD.

According to an Analog Devices article early forms of the technique were developed as far back as the 1940's.

 

RE: DSD [A Reprise]..., posted on September 11, 2014 at 22:33:04
Jaundiced Ear
 Hey Storris!

Welcome to the looniest bin in the entire asylum!

Your own observations are pointing out the reason for a lack of internal PC DSD devices. People only create products if they think they can sell. At a conservative guess, I'd say 99.999% of the market out there is perfectly happy with regular old PCM encoded audio. Indeed, for perhaps 99.9% of the market, PCM encoded audio is overkill. Note the profusion of, nay, dominance of lossy formats in the market.

In the modern market, DSD is at best an afterthought, and only thought about by those who cannot believe that PCM is really all they need to enjoy music. This class of consumer is willing to go to extraordinary lengths to preserve the "purity" of the signal and would no more think of putting a DSD DAC inside of a PC than they would of pushing in their tweeter domes. The OEMs who sell to this tiny group must of necessity be extraordinarily sensitive to the prejudices of their intended market so you are about as likely to see a DSD sound card set up to run in a PCI-E slot as you are to see 5 star restaurants selling hamburgers, however much they may overprice them.

All the Best, and again, welcome to the Asylum! Be sure to check out some of the other fora as well!

JE

 

Here's a modest sound card..., posted on September 11, 2014 at 22:56:46
Jaundiced Ear
 that seems to shrug off the harsh environment inside a PC.

Now come on, in all honesty, where in that battery of measurements does it become apparent that the component under review is in "the harsh environment inside a PC?"

JE

 

RE: Here's a modest sound card..., posted on September 11, 2014 at 23:33:16
Storris
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Location: London
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 ...that card uses a DSD capable chip!

The point is well made. Even my integrated audio produces clean and crisp, analogue audio signals, and that's buried in the middle of the motherboard.

Not to mention the vast array of analogue TV and FM tuners available for the PCI bus, which cope perfectly with the environment.


 

LOL!!, posted on September 11, 2014 at 23:57:51
Jaundiced Ear
 Boy, do you have a lot to "learn" about PC audio! Park your common sense at the door please! Ignore all the information your ears present to your mind! Forget everything you learned in school about logic! Who cares if there are literally thousands of intelligent, skilled and educated people out there pushing on a daily basis the boundaries of audio fidelity? There are a dozen or so folks here who will explain to you how you and everyone else are all wrong, and some of them will even be happy to sell you multi-thousand dollar solutions to problems you never even knew existed!

Your trip down the rabbit-hole hasn't even begun!

JE

 

RE: This seems easy, posted on September 12, 2014 at 00:00:14
Storris
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 That's the box I was referring to that would fit entirely into the space in my PC.

The cost isn't the driving force here, although obviously important.

What is important is that the industry seems to have decided to provide the least optimal solution.

In Loki's case the entire chassis, along with a considerable part of the board, could be cut by utilising the PCI bus.

 

RE: LOL!!, posted on September 12, 2014 at 00:28:51
Storris
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Location: London
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RE: This seems easy, posted on September 12, 2014 at 00:32:37
Storris
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Location: London
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 What is also important, is that the market has accepted the least optimal solution.

 

Now don't get me wrong..., posted on September 12, 2014 at 00:41:47
Jaundiced Ear
 the people here may be delusional, but they are not stupid. I'll freely admit that most of them are smarter than me.

Have you ever boxed? Going against an inferior opponent may be satisfying, but you learn little. Going against a superior opponent may be embarrassing, and end up with you on your ass, with a bloody nose, but you will learn a lot.

I've learned a lot from posting here. But before I started posting here I'd already learned how to hang on to my wallet.

Have fun here, but make sure to keep an eye on your wallet!

JE

 

RE: Now don't get me wrong..., posted on September 12, 2014 at 03:05:49
Storris
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Posts: 32
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 You'll get no arguments from me on the education and fun to be had, and thanks for your input.

 

RE: This seems easy, posted on September 12, 2014 at 03:26:35
Mercman
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Posts: 6581
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 It all depends what your goals are. Sure, you can have enjoyable music with solutions proposed by JE. But for those who desire a "high end" audio experience, using a sound card in a computer is not going to deliver commensurate sound quality with associated high end components.

It's your choice, but don't be deluded by the people who stink up this forum.

 

RE: This seems easy, posted on September 12, 2014 at 04:27:32
Storris
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 The goal is bit-perfect, high-res, analogue, output from the PC.

The PC is capable of all 3 of those things, just not currently at the same time.

 

Have you asked Schiit,, posted on September 12, 2014 at 06:46:47
G Squared
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 if they would make you one as you describe. It would not work for my application since I use a laptop.

I have a Schiit Bifrost DAC and have not ventured into DSD.
Gsquared

 

RE: DSD [A Reprise]..., posted on September 12, 2014 at 07:09:23
Tony Lauck
Audiophile

Posts: 13629
Location: Vermont
Joined: November 12, 2007
 "Sure Sony invented "DSD". They are credited with developing and bringing practical application of sigma-delta modulation ADC to the audio entertainment industry, and they called it DSD."

I would not confuse inventing with marketing a technology for a particular application.

Steve Jobs and Steve Wozniak brought 1 bit audio technology to the telephone hacking community with the Apple II (or maybe Apple I) around 1980. The built in speaker was connected to a software controlled flip flop. Generating the two tones for a Blue Box required a form of pulse density modulation, since at least three level signaling was needed to sum the output of two square waves.


Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar

 

RE: Now don't get me wrong..., posted on September 12, 2014 at 07:44:32
AbeCollins
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 You're welcome to use an internal sound card or onboard audio if you wish. Most here prefer the flexibility of using an outboard DAC. I've tried a number of them and they all sound a little different.

 

RE: DSD [A Reprise]..., posted on September 12, 2014 at 09:44:13
AbeCollins
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 OK, you make a valid point. We should bring up 'prior art' and give credit to Gottfried Wilhelm Leibniz who invented the modern version of our binary number system. ;-)

I was hacking phones as a kid in the 1970's. I bought a surplus Ma Bell touch tone decoder board about a foot square in size. It had multiple passive LC tuned circuits on it (audio filters) to decode tone pairs and discrete logic using bipolar transistors at the output.

I later used the LM567 tone decoder PLL chip, 8 of them with additional logic gates to decode all the necessary tone pairs for a 16 key pad. These were OK but the passive RC circuits weren't all that stable over temperature and had to be re tuned (10 turn trimpot) frequently. Not practical for the application I had in mind.

And then a couple companies came out with single chip DTMF decoders using a quartz crystal frequency reference and 4-bit binary output that could decode all 16 tone pairs. I added some logic (74154 series decoder and 7474 series Dual-D flip-flops) to decode touch-tone sequences to latch relays ON or OFF to control devices over the phone or via radio.

It's all done in software now using micro controllers - PIC and others. ;-)

I wasn't really 'hacking' phones. We were building 'auto-patch' systems using FM radio repeater stations located high atop tall mountains tied to telephone lines.

Back in the days before cellphones existed, we were making phone calls from our cars or hand-held FM radio transceivers for FREE. A touch tone sequence from a 16-key pad was sent through the radio's mic to 'bring up' the auto-patch on the mountain. Auto-patch was the automated linking of the radio system to the phone line. Once we heard dial-tone on the radio, we could use the same touch tone pad to dial a phone number. At the end of a call another touch tone sequence was sent to hang-up the line. A watchdog hang-up timer was used to hang-up the phone line in case we lost radio contact and couldn't hang-up the line remotely.

These projects were time consuming and costly so we had a number of people involved working on different aspects of the system. They were called HAM Radio clubs. We collected dues to pay for equipment, electricity, and the phone line. Only members had the touch-tone sequence for accessing the auto-patch system. There were other safeguards as well.



 

"goal is bit-perfect, high-res, analogue, output from the PC" - whose goal is that?, posted on September 12, 2014 at 10:44:00
carcass93
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Posts: 7181
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 Not mine, for sure - and not of those who actually compared various solutions in resolving system.

I actually - personally - compared ASUS Essence STX (helped by feeding from external linear PS) versus USB output to W4S uLink to a quality external DAC, using optimized for audio playback machine, in sufficiently resolving system.

ASUS satisfies all your criteria - bit-perfect, high-res, analogue. Do you think the two options above sound the same?

 

"stink up this forum" - well put, if too mildly for my tastes., posted on September 12, 2014 at 10:47:16
carcass93
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 We're seeing his actual excrements everywhere, not just smell the stink, IMO.

 

RE: "stink up this forum" - well put, if too mildly for my tastes., posted on September 12, 2014 at 11:34:47
Mercman
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Posts: 6581
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 The man has been here for well over a year and has learned nothing. The same simplistic questions and comments over and over again. Worse than that, he knows that there are no measured verifiable answers to his demands, but enjoys pissing on us regularly.

The funniest comment he has made to me was that since I'm a podiatrist, what qualifies me to write a review?

He won't share with us what his system is or any particulars.

Why does he feel the need to turn an audiophile forum into a Hydrogen Audio forum?

I guess he does have a purpose in this forum. We have someone to dump on. But it does get old very quickly.

 

RE: Have you asked Schiit,, posted on September 12, 2014 at 11:35:23
Mercman
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 They seem like great products for the money.

 

Yeah... besides that one undeniable purpose, he serves another - providing "diversity"., posted on September 12, 2014 at 11:43:50
carcass93
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 Or so the moderator thinks.

 

RE: "goal is bit-perfect, high-res, analogue, output from the PC" - whose goal is that?, posted on September 12, 2014 at 12:37:12
Storris
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 The ASUS STX is capable of 24/192, but so are most integrated chips.

I think Hi-Res probably needs to be qualified as DSD/DXD currently, or whatever the highest achievable bit-rate is at any given time.

To qualify even further, given that my audible range is ~15Hz to ~16kHz, would it be too much do define full Hi-Res, as 16000kbps (16Mbps)?

I appreciate your goal is not bit-perfect analogue from the PC, but surely you still want bit-perfect?

Quality of particular sounds, colours of tones, shape of the soundstage and other subjective descriptors... these are all artefacts of a given system, distortions if you will, of the original analogue signal, which can be moulded by the listener at their convenience, and to their taste.

But surely all listeners have 2 common goals in sourcing high quality, closest to original as possible material, coupled with a bit-perfect analogue output?

 

RE: This seems easy, posted on September 12, 2014 at 12:39:52
Storris
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 I haven't asked Schiit , but I have put the proposal to PS Audio.

 

RE: "goal is bit-perfect, high-res, analogue, output from the PC" - whose goal is that?, posted on September 12, 2014 at 13:02:55
Storris
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Posts: 32
Location: London
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 Woah, let me just back up a bit here...

So DSD64 gives a bit-rate of 2.8Mbps?
24/192 gives " " 4.6Mbps?

So I currently have better than DSD audio definition, which is cool. However...

According to my Full-Res ambition we're looking for something around DSD384 at ~16.8Mbps or Double-DXD = 24/666kHz (spooky), double DXD, to qualify as Full-Res-Audio.

Is this even close to sensible?

 

RE: "goal is bit-perfect, high-res, analogue, output from the PC" - whose goal is that?, posted on September 12, 2014 at 13:07:14
Storris
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 Just want to highlight that 24/666 is equivalent to 16Mbps exactly.

..and that writing "Double-DXD" twice was unintentional.

 

RE: "goal is bit-perfect, high-res, analogue, output from the PC" - whose goal is that?, posted on September 12, 2014 at 14:23:46
AbeCollins
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"Just want to highlight that 24/666 is equivalent to 16Mbps exactly.

And you want 2-channels for stereo, right? Make that 32Mbps.

CD quality is 16/44.1 which is 705.6Kbps, x2 = 1.4Mbps

24/192 'hi-res'has a bit-rate of 4.608Kbps, x2 = 9.2Mbps

Given the state of electronics today do you think you will hear the difference between 24/192 vs 24/666? And do you want more than a couple albums at that resolution to fit on your smartphone? That's about 1GB per 4-minute tune. ;-)



 

RE: "goal is bit-perfect, high-res, analogue, output from the PC" - whose goal is that?, posted on September 12, 2014 at 15:39:40
Storris
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 Just as the difference between 1080p and 360p video is obvious, Hi-Res audio should be just as perceptible (roughly a factor of 3, the same as 24/192-24/666). I obviously can't speak definitively, but human physiology says it's a certainty. I guess we'll just have to wait to find out.

I'm not sure I want or need that level of resolution on my smartphone, the primary concern is the PC where I do the majority of my listening... but I wouldn't turn it down.

And fortunately, SanDisk has just released a 512GB SD Card with 95MB/s transfer rate. That's plenty of room for several days typical mobile listening. As I said at the top storage capacity is not an issue, on any platform.

http://www.sandisk.com/products/memory-cards/sd/extremepro-sdxc-sdhc-uhs-3/?capacity=512GB

 

Hope they make it for your application nt, posted on September 12, 2014 at 16:10:44
G Squared
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 Nt
Gsquared

 

RE: "goal is bit-perfect, high-res, analogue, output from the PC" - whose goal is that?, posted on September 12, 2014 at 17:25:02
Jaundiced Ear
 "Just as the difference between 1080p and 360p video is obvious, Hi-Res audio should be just as perceptible (roughly a factor of 3, the same as 24/192-24/666)."

In this case the base level format had output that was nowhere near the limitations of the human eye. 1080p was much closer and so looked better. When you get to the so called "retina display" levels of resolution, you're pretty much up against the limitations of the human eye. Tripling the resolution of a retina display is not going to give you a three times better picture.

You'll find the core dispute in this forum is between those who argue that human hearing has some limitations to it and therefore at some point further refinement to the audio signal is inaudible while others argue that the limits of human hearing are still unknown and that every change to a system, however subtle, may be audible. The first group argues that for changes to be audible they should also be measurable, the second group argues that the ear can hear changes that are not measurable and therefore hearing trumps measurement. The first group thinks that "Redbook CD" is pretty darn close to the limitations of human hearing and so is "transparent," while the second group thinks Redbook is not close enough to the unknown limits of human hearing to be considered transparent, therefore they want even more elaborate formats and greater signal density, so called "Hi-Rez" music. The first group thinks this is a golden age of audio because even modestly priced equipment can be "transparent," while the second group also thinks this is a golden age of audio because however pricey and fancy the piece of equipment under consideration it can always be further improved by additional tweaking.

JE

 

I am enjoying the Bifrost. Nt, posted on September 12, 2014 at 17:58:25
G Squared
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 Nt
Gsquared

 

RE: "goal is bit-perfect, high-res, analogue, output from the PC" - whose goal is that?, posted on September 13, 2014 at 01:03:47
Storris
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Posts: 32
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 There's an important difference between ppi and resolution when it comes to video/images. High PPI retina-type displays simply allow higher definition on smaller screens, rather than being a measure of High Definition per se.

Whether or not one can hear the difference between DSD64/24@192 and DSD384/24@666 should just require the software to produce the required frequency for the required time.

That software would allow each of us to determine the Highest-Res distinguishable on an individual basis with an objectively definitive result, provided the use of equipment capable of translating the analogue signal of course.

And I'm almost certain that I can find that software on the interwebs. I'll be back with results after MotoGP.

 

RE: "goal is bit-perfect, high-res, analogue, output from the PC" - whose goal is that?, posted on September 13, 2014 at 03:21:03
Storris
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 I tried NCH Tone Generator but it is pretty limited, only allowing to test maximum @ 1Khz due to inability to adjust tone length below 1ms, or to adjust sample rate above 192Khz. Still works good for what it does do.

This looks more promising for high-end systems - http://www.esseraudio.com/en/test-tone-generator-windows-software-generate-test-signal-sine-pink-noise-crest-factor/ttg-dl-en.html - It's heavy overkill for me, but with sample rates up to 24/384, it should be perfect for testing resolution and tuning the highest-end set ups.

 

Redbook Proven to be not transparent, posted on September 13, 2014 at 10:06:18
Tony Lauck
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 "The first group thinks that "Redbook CD" is pretty darn close to the limitations of human hearing and so is "transparent"

The better recording engineers and subjective audiophiles have known for a long time that high resolution digital audio is needed to transparently reproduce what humans can hear. These people have simply used their God given senses to note the obvious. For a long time a different gang of pseudo scientific hobbyists, calling themselves "objectivists", have conducted experiments that they mistakenly claimed showed that the "subjectivists" were delusional and that nothing better than 44/16 PCM was necessary for transparent reproduction of music.

Recently, the ground on which the objectivists stood has been washed down the river to the sea. The linked thread (and some threads linked off this thread) discuss how this was done.

Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar

 

Subjective objectivity, posted on September 13, 2014 at 12:46:51
Storris
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Location: London
Joined: September 8, 2014
 There seems to be passionate support for both 'sides' of this discussion. However, I'm not prepared to accept the word of one unknown individual over my own experiences on any subject, not even when his findings agree with my own. And given that we have the tools to do our own digging, and that the results of such digging will be unique for each of us, I don't see that we should even contemplate it.

There's an "I owe an apology to BIS" thread in the HiRez Asylum, after a few public trashings of a new piece of equipment the OP had his hearing tested and discovered that his ears were at fault. For many people there is going to be no benefit gained from Higher-res audio, either due to physical impairment of unsuitable equipment.

I have an image of speakers topped with various colours of lights that represent the frequencies a listener can't hear.

 

RE: Subjective objectivity, posted on September 13, 2014 at 13:31:09
Tony Lauck
Audiophile

Posts: 13629
Location: Vermont
Joined: November 12, 2007
 The necessary files and tools are easy to download and run. Several people have reported positive results, e.g. ability to pass blind tests 20 out of 20 times repeatedly, except for an occasional time when a nearby dog barked.

Regardless, the claim that no one had ever passed such a blind test no longer stands. Of course the deniers remain unconvinced. But this just proves that these "objectivists" were not and are not objective. Reading all the threads all the way through says a lot about the personalities.


Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar

 

RE: Redbook Proven to be not transparent, posted on September 13, 2014 at 14:03:02
Jaundiced Ear
 Hi Tony!

Golly! Do I have to read that entire 1,000+ post thread?

"Recently, the ground on which the objectivists stood has been washed down the river to the sea."

I'm not sure I'd be willing to declare that just yet. I've only read a few pages of your link so far. First of all I'd like someone to check the provenance of the "smoking gun" files to make sure there wasn't an issue in preparing them. Secondly, are we testing the transparency of 16bit, 44k files or the audibility of the conversion of files to that format? Finally, if the test is legit and the results correct, then that would be insanely cool! Could these results be expanded to more general musical files than some homemade files of jingling keys?

All the best and looking forward to reading more of that thread!

JE

 

Food for thought, posted on September 13, 2014 at 14:09:19
Jaundiced Ear
 Hi Storris!

I hate to post this link because of the outrage it often engenders, but you may find it interesting.

JE

 

RE: Redbook Proven to be not transparent, posted on September 13, 2014 at 14:19:40
Tony Lauck
Audiophile

Posts: 13629
Location: Vermont
Joined: November 12, 2007
 You have to read all three threads. More like a total of 4000+ posts! You will see all of your questions, and more, discussed and debated ad nauseum. There are musical files too, from AIX records.

Plus you have to download the audio selections and listen to them. To understand some of the discussions it is also useful to run the ABX software and see if you can hear differences or artifacts from the test software itself.


Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar

 

RE: Food for thought, posted on September 13, 2014 at 14:59:22
Storris
Audiophile

Posts: 32
Location: London
Joined: September 8, 2014
 Thanks for the link. I've been thinking about the enormous range of frequencies available through 24/xxx myself.

I don't see the benefit of recording them, storing them or trying to reproduce them at the speaker.

Any harmonic effects in the studio have already had their effect on the audible range, and that was recorded. Recording the high frequencies and reproducing them through the speaker;
allows them to have a second harmonic effect (this is equivalent to 'feedback' isn't it?),
makes it more difficult to produce the audible range,
and disrupts the audible range's ability to produce its own natural harmonics in the listening room.

-------------------
This being the case, for most of us sample rates above 32kHz are pointless, and as for DSD... So, a new objective...

16Mbps from a 32kHz sample rate means a 500-Bit Depth!
-------------------

Reconsidered...

Ok, so 16Mbps is the maximum amount of auditory processing, listening, I can do at any given time. Meaning that if 16Mbps of music (i.e. a constant 16kHz tone at highest perceivable volume) was presented to my ears, I would be incapable of responding to any other sound, or my brain might explode, maybe. 16Mbps is not desirable.

Any sample rate above 44.1kHz is wasted, in every sense.

I've only got 16bits, but they provide 65,000 points of resolution.

Ok, JE, I think I've got most of it. A couple of things I'm not clear on, such as how a given sound is digitised, say a 1 sec beat and decay of a C# bass drum. But I guess that would apply to all A/D signals (FM, TV etc...) and isn't strictly relevant.

 

RE: Subjective objectivity, posted on September 13, 2014 at 15:06:06
Storris
Audiophile

Posts: 32
Location: London
Joined: September 8, 2014
 Sorry, I'm just not sold on the validity of self-administered blind tests.

...besides which, I don't think blind tests of any sort are necessary.

The tone generator method I described earlier will give a much more accurate result of what resolution the listener or his equipment can achieve, while fidelity can be measured via spectrograph.

This might all spoil what is supposed to be an enjoyable past-time though, so I'm not recommending this level of analysis for anyone who enjoys listening more than fiddling.

 

RE: Food for thought, posted on September 13, 2014 at 17:30:00
Tony Lauck
Audiophile

Posts: 13629
Location: Vermont
Joined: November 12, 2007
 "I hate to post this link because of the outrage it often engenders"

No outrage here. I think the problem is at worst a matter of willful ignorance. I think the author is a fool whose bogus world view has recently been shown to be inconsistent. If he is a person of integrity he faces a rude awakening. I feel sorrow, not outrage, in the face of ignorance.

Now if you want to talk about the prices of hires downloads, that might be different. There is no justification for a huge premium on high res versions of new releases as there are insignificant extra costs involved.

Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar

 

"Outrage" is interesting word. Do I feel outrage, reading that non-experiential crap?, posted on September 13, 2014 at 19:04:38
carcass93
Audiophile

Posts: 7181
Location: NJ
Joined: September 20, 2006
 No, not at all.

The same way, I didn't fell "outrage", when some years ago I was passing by an institution in Philly, called something like "Institute for Mental Health and Mental Retardation", and seeing some of the clientele wandering around and sitting on the stairs. Anyone who's been to Philly knows that half of the population there could potentially use the services of that institution, but those who (forced to) actually do - they are "crème de la crème", so to speak.

Did I try not to look? Sure - not too successful, though. That's why I read that article, too.

 

RE: Food for thought, posted on September 14, 2014 at 21:46:59
Storris
Audiophile

Posts: 32
Location: London
Joined: September 8, 2014
 JE,

Would it be fair to say that recording in DSD, or at the highest possible sample rate, would produce an orders of magnitude more accurate digital master than the alternative, and that (given the correct equipment) D/A conversion of such recordings will produce a much more accurate analogue output?

Presuming you concur...

Any analogue signal is continuous, but the digital signal from which it was made is not, leaving gaps between samples which should be observable on a spectrograph with sufficient resolution. At any sample rate which is below the speed of analogue, these gaps will be transferred to the analogue as silences (or noise/whatever).

Because the analogue signal is travelling at something approaching the speed of light and because of the listener's auditory capability/limitations of his equipment, none of these gaps will be heard by the listener, unless transferred as noise and left unfiltered (if transferred as noise, one would presume that they would be of a constant nature and therefore identifiable and susceptible to filtering, leaving the gaps as silence).

What this means is that the speaker is playing a broken signal. Where there is a silence, the speaker merely waits an imperceivable amount of time (0.000021secs for 48kHz)[¹] for the next bit of playable signal to arrive.

However, where there is a more accurate analog signal, the gap will be much shorter and will be followed by a signal that would not have been reproduced with a lower sample rate.

Similarly, a RedBook CD pressed from a High-Res master will contain different information to a RedBook Cd made from a Lower-Res master of the same track.

There will be audible and measurable differences between higher and lower-res music, even if you aren't actually listening/playing back at a higher resolution.

Note 1:

Whether this is actually an imperceptible silence should easily be tested with the correct software, introducing the same gap into a constant tone. My calculations a few posts up suggest that 0.00005 is the shortest perceivable time for a 20kHz signal, and 0.001 for a 1kHz signal, the inverse should also hold with the equation 1/xHz providing the minimum audible time for a given frequency. i.e. 0.000021secs of silence is less than half of what is needed to produce an audible change at 20kHz, and is thousands of times shorter than can be perceived at 1kHz.

 

RE: Food for thought, posted on September 14, 2014 at 22:30:32
Jaundiced Ear
 Hey Storris!

"Would it be fair to say that recording in DSD, or at the highest possible sample rate, would produce an orders of magnitude more accurate digital master than the alternative, and that (given the correct equipment) D/A conversion of such recordings will produce a much more accurate analogue output?"

Actually, I don't concur with that. Firstly, I know nothing about how DSD works, so I can't offer any opinion about that. I know next to nothing about how PCM works, but what little I do know makes me think that increasing the frequency of the sampling rate does not "move the samples closer together." Instead it adds higher frequency data to the signal. You already picked up the information between 20Hz to 20kHz when you recorded at 44kHz. Recording at 192kHz is only going to add in high frequency harmonics to the original 20Hz to 20kHz signal. Similarly, increasing the bit depth does not make the samples smoother, instead it lowers the noise floor.

I'm sure others will be happy to leap in and correct me if I'm wrong!

Further, what comes out of a DAC is analog. Any digital "gaps," (which are there because of the nature of digital but which don't matter because in a bandwidth limited system there is only one, unique analog waveform that correctly corresponds to the digital information) have been filtered out in the conversion to analog. It is an analog signal passing through your amplifier and from there to your speakers. The signal is not "broken" into discrete pieces.

Even if it was, your speakers are not quantum devices. They do not move from one state to another with no intervening states. Even if you could build a quantum speaker, they would have to push against ordinary air which would thereafter propagate an analog wave.

By the way, another link to that guy who is roundly despised here.

JE

 

RE: Redbook Proven to be not transparent, posted on September 14, 2014 at 22:32:03
Jaundiced Ear
 Hi Tony!

"You have to read all three threads. More like a total of 4000+ posts!"

Yeow! You're going to have to give me some time for that.

JE

 

RE: Food for thought, posted on September 14, 2014 at 22:40:30
Jaundiced Ear
 Tony:

"I think the author is a fool.."

Your concept of "fool" must vary widely from mine, and must encompass a lot more people.

"Now if you want to talk about the prices of hires downloads, that might be different. There is no justification for a huge premium on high res versions of new releases as there are insignificant extra costs involved."

Again our opinions differ. I see prices as market place signals. I don't think prices have anything to do with the underlying costs of the product.

JE

 

RE: Food for thought, posted on September 14, 2014 at 22:58:36
Storris
Audiophile

Posts: 32
Location: London
Joined: September 8, 2014
 I'll go and read up some more on exactly what gets sampled and when. I'm basing this on the premise that everything gets sampled, and increasing sample rate also increase sample range.

I appreciate the analogue is a single continuous entity but the breaks do exist, in the digital at least. If the DAC is knitting the seams together, or filling the intervening space with algorithmically estimated data, the result is an analogue signal that could be improved, depending of course on what gets sampled when. I'm off to do some more reading.

Regards

 

RE: Food for thought, posted on September 14, 2014 at 23:06:56
Jaundiced Ear
 Hey Storris!

Remember, we're dealing with bandwidth limited signals. That's one of the keys to digital: we're not dealing with infinite bandwidth, but only a limited set. The theory does sound crazy, but you don't even think about it when you hear voices coming out of your telephone or watch a TV show.

JE

 

RE: Food for thought, posted on September 15, 2014 at 01:33:46
Storris
Audiophile

Posts: 32
Location: London
Joined: September 8, 2014
 Thanks for the links by the way, very informative stuff...

I get what's being said, and think it sounds perfectly sane when applying it to a constant tone, but not with music. As far as I can tell, there is no physical way that 48,000 images of a thing can give you a picture as complete or accurate as 3,000,000 when that thing is changing at an infinitesimal level, constantly. Details will be missed.

This is true for video and still images in digital and analogue, even eyesight, so how not for A/D/A sound?

Anyway, I've put some questions up on his Wiki.

EDIT

Unless the Bit Depth is involved? 16Bits/MoreFrequencies = MoreFrequencies per Bit = Less detail than 16Bits/FewerFrequencies. But Bit Depth is Dynamic range...?

Also,

In response to other questions on the video's wiki page, he or someone at least, has written this;

"Does this mean that we should have better output if we increasing the sampling from 44.1KHz to 192KHz?"

"You'll have exactly the same 20 kHz sine wave at 192 kHz as 44.1 kHz... Only two sample points are needed to perfectly recreate a sine wave... --Leorex"

"It's counterintuitive, but try and think of it like this. You know the input signal (analog) is band-passed to 20kHz, so there are no frequencies higher than 20kHz to be reconstructed. Now look at the 2.2 samples per period; try and draw a continuous line through all the samples without using any frequencies above 20kHz. So in fact, there is only *1* solution for the line you draw through the sampling points. You can 100% recreate the original analog signal from the sampling points. You will not get better output by increasing the sampling rate to 192kHz, because you have already reconstructed 100% of the signal. -- Nhand42"

To my mind these answers only suffice for constant frequency/amplitude signals. When you add a constantly changing frequency/amplitude signal, the ability to "draw a continuous line through all the samples" will become much more difficult. We are not trying to recreate a Sine Wave. Try illustrating an orchestral piece with his lollipop diagram.

 

RE: Food for thought, posted on September 15, 2014 at 04:05:30
Jaundiced Ear
 Sorris:

"I get what's being said, and think it sounds perfectly sane"

No, if you got what was being said you would not think it sounds perfectly sane.

"I get what's being said, and think it sounds perfectly sane when applying it to a constant tone, but not with music. As far as I can tell, there is no physical way that 48,000 images of a thing can give you a picture as complete or accurate as 3,000,000 when that thing is changing at an infinitesimal level, constantly. Details will be missed."

You think this because you have no understanding of how digital audio works. Perhaps this will help.

Way back when, people wanted to record analog events around them (people speaking, or making music}. They only had analog tools so they used them. These analog tools created a one-dimensional analog wave form that people could fiddle with and send over the air, or impress onto vinyl, or use to magnetize iron paritcles glued onto tape. It didn't really matter. Those formats were all used to create analog waveforms that were then amplified by amplifiers and used to drive speakers.

The crucial point to remember is that a signal consisting of a one dimensional, varying voltage, was used to capture and to simulate the sound of music or other noises.

Digital Audio is NOT a description of reality. Digital Audio is a description of the one dimensional, varying voltage that, until now, has been used to simulate sound and music.

So long as you keep trying to use "Digital Audio" to describe "reality" you are going to go insane. Only once you get the idea that "Digital Audio" is only useful to describe "recorded audio" will you get to be comfortable here.

All the Best!

JE

 

RE: Food for thought, posted on September 15, 2014 at 04:30:48
Storris
Audiophile

Posts: 32
Location: London
Joined: September 8, 2014
 "varying voltage"

Two words that have fixed everything, cheers.

 

RE: Food for thought, posted on September 15, 2014 at 05:04:01
Jaundiced Ear
 You just needed a new perspective. Just as there are no "quantum speakers" neither are there any "quantum microphones." When you think about it, is there any part of recorded audio, digital or analog, that can't be attributed to a voltage variation?

JE.

 

RE: Food for thought, posted on September 15, 2014 at 05:21:15
Tony Lauck
Audiophile

Posts: 13629
Location: Vermont
Joined: November 12, 2007
 I understand. As you undoubtedly know, the economic term is price discrimination. For a few years I was responsible for a line of computer peripherals that interconnected mini computers to phone lines. We took one product and through a minor configuration change turned it into two products with almost a 2x price difference. That way we could sell a cheap large version to universities and an expensive small version to Ma Bell. That way I could keep the people who were marketing to universities and the people marketing to the phone company both happy with a "reasonable price". I thought this was a great idea, after all it paid my salary. :=)

Blue Coast Records has a huge price premium on their DSD files, but for the past year they have been putting all formats on a single low introductory price for a few weeks. Eclassical.com does something similar with their new releases and there is only a small premium on their "daily specials".

Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar

 

"You have to read .... 4000+ posts" - or, he could assemble a reasonably resolving system, and listen., posted on September 15, 2014 at 08:44:07
carcass93
Audiophile

Posts: 7181
Location: NJ
Joined: September 20, 2006
 Something tells me it's going to be neither.

 

RE: Redbook Proven to be not transparent, posted on September 15, 2014 at 11:57:33
Tony Lauck
Audiophile

Posts: 13629
Location: Vermont
Joined: November 12, 2007
 For full appreciation it will be also necessary to download test software and the test files and listen to them. You can have an extension. :-)


Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar

 

RE: Food for thought, posted on September 15, 2014 at 12:37:33
Tony Lauck
Audiophile

Posts: 13629
Location: Vermont
Joined: November 12, 2007
 The theory is that the signals are bandwidth limited. Theory is only an approximation to reality because a finite duration signal can only be approximately band limited. For a system to work well in practice there must be a significant guard band between the audio signals and the theoretical bandwidth determined by half the sampling rate.

Also, just to set things straight from another post in this thread, if one has a sine wave it is not sufficient to sample it twice per cycle. One must sample it more than twice. (To see this, consider that the two samples might just have hit the positive and negative zero crossings of the sine wave.) If there are only slightly more than two samples, then the filter has to be very sharp, e.g. "ring" for a very long time, to figure out what the original sine wave was.



Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar

 

You can go ahead and edit your posts, as posting to yourself isn't kosher...*, posted on September 16, 2014 at 09:31:26
Chris Garrett
Bored Member

Posts: 16674
Location: Miami, Florida
Joined: October 9, 1999
Contributor
  Since: June 19, 2000
 *



 

RE: DSD [A Reprise]..., posted on September 17, 2014 at 08:23:23
Yuri Korzunov
Manufacturer

Posts: 10
Joined: September 14, 2014

I want share my experience of creating 1-bit sigma delta modulator
Software developer of HD audio file converter

 

RE: DSD [A Reprise]..., posted on September 17, 2014 at 09:44:01
Storris
Audiophile

Posts: 32
Location: London
Joined: September 8, 2014
 Well Yuri, thank you.

I have downloaded and used your converter. Can I publicise this on another forum?

I think it is a part of the answer to the questions I have.

Do you know the Directstream DAC? Is your conversion similar to what is being done there?

 

RE: DSD [A Reprise]..., posted on September 17, 2014 at 11:20:45
Yuri Korzunov
Manufacturer

Posts: 10
Joined: September 14, 2014
 > Can I publicise this on another forum?
Yes - with link to original article (or with author name), please.

> Do you know the Directstream DAC?

Models? As examples "Zodiac Platinum DSD DAC", "Ayre Acoustics QB-9 DSD DAC", etc. Exists "native DSD" DAC and "DoP DSD" DAC. DoP is "DSD other PCM" mode transfer of DSD audio data via USB.

> Is your conversion similar to what is being done there?
DSD is delta sigma modulated audio stream 1 bit/DSD64(2.8 MHz)/DSD128(5.6 MHz). As manufacturer I can't use terms "DSD" or "Direct Stream Digital" due it's registered trademarks by SONY.
Software developer of HD audio file converter

 

RE: DSD [A Reprise]..., posted on September 17, 2014 at 12:05:55
Storris
Audiophile

Posts: 32
Location: London
Joined: September 8, 2014
 The Directstream DAC is linked below. The design up-samples all digital inputs (DSD & PCM) to 1Bit/28Mhz and outputs as 1Bit/5.6Mhz, using FPGA.

I appreciate your tool is not the same, but would the conversion to .dsf be a similar process?

 

RE: DSD [A Reprise]..., posted on September 17, 2014 at 12:53:19
Yuri Korzunov
Manufacturer

Posts: 10
Joined: September 14, 2014
 DirectStream DAC use sigma delta modulation for converting input PCM data to 1-bit.

If we simply upsample multibit PCM and truncate bit-depth to 1, we get high quantisation noise level (depend on sampling rate).

Sigma delta modulation allow shift most energy of the noise to inaudible frequency range.

DirectStream DAC audio data format = DSD = DSF audio data format = sigma delta modulation with sample rates 44100 Hz x N


Software developer of HD audio file converter

 

RE: DSD [A Reprise]..., posted on September 17, 2014 at 16:50:28
Ted Smith
Manufacturer

Posts: 10297
Location: Seattle
Joined: December 29, 2000
 Howdy

The block diagram is a similar formulation of a sigma delta as I like to use when explaining DSD. In so far as both have sigma delta modulators our approaches are similar - but virtually anything that produces a single bit bit stream will have a sigma delta modulator. I've chosen to upsample all PCM inputs to 128 x 44.1k (via 640 x 44.1k) rather than having two output rates (since the hardware can only be optimized for one.)

Also with audio sigma delta modulators the devil is in the details.

-Ted

 

RE: DSD [A Reprise]..., posted on September 17, 2014 at 20:20:33
Yuri Korzunov
Manufacturer

Posts: 10
Joined: September 14, 2014
 > Also with audio sigma delta modulators the devil is in the details.

Fully exactly! :)
Software developer of HD audio file converter

 

RE: DSD [A Reprise]..., posted on September 20, 2014 at 10:21:33
Storris
Audiophile

Posts: 32
Location: London
Joined: September 8, 2014
 Hi Yuri,

I am getting a lot of clipping when converting .dsf/.dff to FLAC. Are there any settings that will help?

 

RE: DSD [A Reprise]..., posted on September 20, 2014 at 11:18:14
Yuri Korzunov
Manufacturer

Posts: 10
Joined: September 14, 2014
 Hi Storris,

Cliping with AuI?

1. You use DSF or DFF?

2. What sample rate of input DSF and DFF?

3. What sample rate of output FLAC?

4. Can you send me source file to dropbox or other?

Best regards,
Yuri
Software developer of HD audio file converter

 

RE: DSD [A Reprise]..., posted on September 20, 2014 at 11:39:01
Storris
Audiophile

Posts: 32
Location: London
Joined: September 8, 2014
 Yuri, thanks for the response. Yes clipping is with Aul,

1. .dff

2. 2.8Mhz

3. All sample rates/bit depth at FLAC and .wav I haven't tested other formats.

4. http://1drv.ms/1raVJIt - is a Skydrive link.

Regards

SM

 

RE: DSD [A Reprise]..., posted on September 20, 2014 at 11:41:17
Yuri Korzunov
Manufacturer

Posts: 10
Joined: September 14, 2014
 Thank you. I will check it.
Software developer of HD audio file converter

 

RE: DSD [A Reprise]..., posted on September 20, 2014 at 11:46:58
Tony Lauck
Audiophile

Posts: 13629
Location: Vermont
Joined: November 12, 2007
 If you don't have a gain control, you need one for converting to PCM formats other than floating point. There is just too much variance in levels with DSD recordings. Existing converters such as Korg Audiogate and Weiss Saracon have this feature.

As a workaround, convert DSD to floating point and then use other software to reduce gain and convert to integer formats.

Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar

 

RE: DSD [A Reprise]..., posted on September 20, 2014 at 11:57:40
Yuri Korzunov
Manufacturer

Posts: 10
Joined: September 14, 2014
 Hi Tony,

I don't use now direct access to DFF (due license to DFF issue). It converted via free decoder to intermediate WAV, after resampled/changed bit-depth, after converted to target format.

To DSF AuI has direct access. DSF opened as 64-bit floating point (double precision) and all processings AuI do in 64-bit floating point.

Thank you for suggestion about volume control, I will think how apply it.

Best regards,
Yuri
Software developer of HD audio file converter

 

RE: DSD [A Reprise]..., posted on September 20, 2014 at 12:02:53
Storris
Audiophile

Posts: 32
Location: London
Joined: September 8, 2014
 Tony,

I have tried using Saracon and Audiogate to convert files but neither seems to work without their proprietary hardware. I have also tried HQPlayer from Signalyst but cannot find the "Convert to..." button. Am I missing something?

 

RE: DSD [A Reprise]..., posted on September 20, 2014 at 12:45:47
Tony Lauck
Audiophile

Posts: 13629
Location: Vermont
Joined: November 12, 2007
 I have used Audiogate to convert to/from DSD. The version that I have requires a Twitter account and "tweets" every time the converter is used in lieu of Korg hardware. The newest version may no longer allow this mode, I don't know.

HQPlayer converts from file formats to the audio device, not to file. There may be a way to capture the ASIO output with some kind of asio pseudo-device, but I haven't tried this. (Note: HQPLayer also has a gain control.)

I haven't tried Saracon. It's expensive and some people don't like its sound quality.


Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar

 

RE: DSD [A Reprise]..., posted on September 20, 2014 at 12:48:12
Storris
Audiophile

Posts: 32
Location: London
Joined: September 8, 2014
 Thx Tony, I've found a usable version of Audiogate.

 

RE: DSD [A Reprise]..., posted on September 25, 2014 at 01:43:41
Yuri Korzunov
Manufacturer

Posts: 10
Joined: September 14, 2014
 Special thanks, Tony Lauck and Storris for info and suggestions.


Now released new AuI ConverteR 48x44 v. 3.8.7 with volume control -40 ... +40 dB and 0.1 dB step (enabled for versions PROduce-R and PROduce-RD).

I while check converting file by Storris to intermediate wav (yesterday finished adding of volume control and fixed bug with BWF WAV files [was detected in free version], generated by Pro Tools). It has max peak levels. I will check way intermediate wav via AuI.

Best regards,
Yuri
Software developer of HD audio file converter

 

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