[goobicii]

   
This is only true on an unfiltered NOS DAC - the image frequencies 'beat' with the signal and cause amplitude fluctuations. Install the appropriate reconstruction filter (most NOS DACs don't do this) to sufficiently attenuate those images and the problem is solved. Its not 'due to insufficient sample rate' as far as I'm aware.

 
but then you are hearing altered signal not original one

 

[Sorrodje]

.

 

[azteca x]

  44.1khz can reproduce constant sinewave only up to about 1600hz,higher and the loudness jumps up and down due to insuficient sample rate
 
yesterday I made experiments,and I make this claim "anything less that 1536khz is lossy format"

 
44.1/2=22.05. Add in some wiggle room for filtering and whatnot and you get up to about 20kHz (20,000Hz) pretty reliably. Of course, I'd love to see a test where you demonstrate that you can hear differences ±0.1dB at 20kHz.
 
The highest sample rate I've ever seen (for PCM) is 384kHz, and it's really just hardware support. There are like two things recorded in that format. No recorder can use the extra high frequencies. And as Lavry pointed out in that whitepaper, there is a time where higher sample rates introduce more distortion.
 
So, how did you do this 1536kHz test?
 
Also, lossy refers to whether or not a format preserves the original RECORDED signal. So if you can take the original WAV and then take a FLAC you bought off of Bandcamp, invert phase on one of them and get a complete null (silence), you have a lossless format. I don't think anyone would claim that there is a lossless recording method. Recording is a game of compromises and always will be.

 

[prot]

Who ******* cares? Really, does it matter? Its every ******* thing inside the Yggy (or any other DAC) matters. The huge transformers, the huge choke, the shunt regulators, SPDIF / USB receivers. Precision DAC chips. EVERY ******* THING matters. The filter is just a part of it. Like the valve train in an engine of a complete car.

many called that filter "special sauce" or something like that and IIRC you did it too ... so I really dont understand what's wrong with someone asking just how 'special' that is ... or why is everyone jumping so hard on that guy for just asking some questions

Anyway, got another burrito Q: guess it is safe to assume that the original filter does still exist as Theta's property .. are there any other current devices using it ? (theta or licensees, or...)

 

[azteca x]

Keith, if it helps: http://www.head-fi.org/t/764787/yggdrasil-technical-measurements

 

[goobicii]

   
44.1/2=22.05. Add in some wiggle room for filtering and whatnot and you get up to about 20kHz (20,000Hz) pretty reliably. Of course, I'd love to see a test where you demonstrate that you can hear differences ±0.1dB at 20kHz.
 
The highest sample rate I've ever seen (for PCM) is 384kHz, and it's really just hardware support. There are like two things recorded in that format. No recorder can use the extra high frequencies. And as Lavry pointed out in that whitepaper, there is a time where higher sample rates introduce more distortion.
 
So, how did you do this 1536kHz test?
.

 
with oversampling,a process that exist only becose the truth is that 44.1khz is too low samplerate
 
what who seen,what records exist or what are current technology limitations is not my point,the point is anything less than 1536khz CANNOT play a sinewave with constant volume in human hearing range 20-20.000 khz.                personaly I hate the Lavry guy,he thinks he is some guru but he is just bad influence becose people who doesnt think and experiment themself follow him blindly,high sample rates are future,Lavry and his BS is just slowing the technological advancement
 
simple,just create sinewave at whatever you consider highest freqency that should be reproduced accurately,lets say 20.000hz,create that sinewave at various samplerates and tell me at what point it becomes perfectly constant in volume without any loudness jumping

 

[AustinValentine]

 @prot - This is the answer to why people are losing their patience with Diamondears. He's essentially been posting the exact same thing in this thread for over a month now without engaging any of the substantial replies that he's received. See inside the spoiler to see what I mean, with posts presented in reverse chronological order [with all of the insulting/bickering/trolling comments in-between removed]: 
 

Spoiler: Warning: Spoiler! [Diamondears Filter Loop Remix

]  
Quote:

You think the Yggy is all about that recycled megaburritto filter to avoid huge costs in designing a better filter?

 

   
Isn't it a fact? And this?--Yggy used the AD R2R DAC chip, because they want to use that old digital filter designed only for an R2R chip. Non-standard/common digital filters are proprietary=expensive.
 

 

And moreso your opinion.

The filter used was designed for R2R 25 years ago. Knowledgeable and experienced, yet just recycle filters from 25 years ago?

 

So if the Yggy didn't have that excellent closed-form filter, it would still sound good, bat?

 
Quote:

Exactly, so you shouldn't get your panties in a bunch too when someone says saying "Sabre/DS chips sucks" sucks big time, especially when it's the filter that's at fault.

So, based on your story, it is that mega-burrito filter that matters, and the reason the R2R/PCM63 is used is because the mega-burrito filter would work on it. Developing a filter takes a while and resources, so why not get or borrow an existing filter (the mega-burrito one, old one, existing and cheap) and use a chip where it could be implemented (the R2R/PCM63)? Genius! So basically we just got back to 1990s DAC technology. Now that's advancement, use technology 20 years ago, put it in a shiny new modern case, and proclaim modern chips sucks big time and that the old one blows (both the chip and California burrito filter). I knew it was the filter.

 

I've never said the Yggy's sound sucks. Again, I just don't like the Sabre-D-S sucks statement when it's not proven that the glare is attributable EXCLUSIVELY to the DAC chip. My suspect is the stock standard digital filter used as this is the common denominator in all the DACs I've heard that I felt are glare-y.

I also said I suspect this thread is related to the Sabre/D-S sucks statement and are used to advertise the Yggy...but I never said Schiit is fooling us in the claimed excellent sound of the Yggy. I fact I said I'd like to try and would likely buy it to hear for myself. The 15-day trial for $115 plus shipping is reasonable for me.
See above. And if that's the case (not saying it isn't), why the need to bash Sabre/D-S DAC chips and proclaim them the primary culprit in present day digital glare?

 

I'm not an EE, but from what I understand (and I'd like to be corrected if I'm wrong on this), the stuff you're saying is in the digital filtering process (oversampling) to remove the noise, not in the DAC chip itself, although I understand that the DAC chip always have stock standard digital filters. This is why I'm saying the culprit is the digital filter process (to remove the noise), not the D-S DAC chip or digital to analogue conversion process itself that went before.

 

Is it possible to create an optimum digital filter/DSP that will remove the glare for the sucky Sabre/D-S DAC chips?

Edit: Or is it better to just use more expensive non-D-S DAC chip to save time and brain matter to develop a digital filter/DSP to remove the glare?

 

He's not being "insulted" (which I'm not, just pointing out things) or criticized for his approach, but for his insulting approach. He's being criticized for bashing others to promoted himself or his products. And he hasn't proven that the glare comes from the D-S chip, my suspect is the stock standard digital filter that comes with the chip.

If it's really not the filter and it's the D-S chip, why not use that same or similar standard digital filter with lots of pre and post ringing together with the AD chip? Why make a custom filter for it? Because the filter plays a big role in the digital-ness or glary-ness of the D-S DAC chips being touted as sucking big time.

 

It wouldn't sell as much because there's no unique-ness necessary for marketing success like the R2R and military-scientific-grade AD chip.
I agree that happens on products for the masses (including food, sadly). But not for higher-end or more top-tier models/products. I like my setup, but I'm open to improvements, and I've heard lots of DAC much much more expensive but sounded worse or the same. But why bash D-S DACs when the fault is in the stock Standard digital filter?

 

How much is each R2R DAC chip? Let's say $250. If in fact they'd present better detail resolution and prevent digital glare, you think audio manufacturers and designers wouldn't budge to get them? Cmon now. And we're not talking here about definition yet (such as bass articulacy) and noise elimination/reduction that over sampling digital filters' main goal is for.

Again, IMHO, the culprit on the digital glare accusation is the Standard digital filter with lots of pre and post ringings, not the D-S DAC chip.

I may be getting the Yggy just for kicks, and keep it if my current opinions gets kicked by it. What digital filters does it have btw?

 

Ok. When I refer to the Standard digital filter, I refer to the Optimal Spectrum filter of Audiolab, not the slow filters. Standard Digital Filter=fast brick wall filter, the standard filter that measures best objectively, the most common digital filter for so long found in most DACs and CD players. The one that has no timing phase distortions nor frequency issues, except pre and post ringings. What should we call this filter?

And even assuming my pet Optimal Transient filters still have pre and post ringings, I'm pretty sure they're minimal or substantially less compared to the Standard digital filter.

But we're getting out of my main point---that you cannot fault the glary-ness on the D-S DAC chip when you're hearing it using the Standard digital filter that has perfect phase and frequency response but has substantial amount of pre and post-ringing not only objectively but also subjectively.

You didn't even mention what filter you used you're basing your ranking on. I don't like bashing D-S DACs, or any DAC chips for that matter, and I strongly suggest account for or review and rank the digital filters used in the market today. I'll be all ears on that.

I suggest further that you check for yourself the Audiolab DACs and see and say for yourself how it measures (objectively) and sounds (subjectively) using the Optimal Transient digital filters, which by the way are slow filters tweaked by the designer that maintains no timing phase distortions.

I'm not sure why digital filters aren't talked about much, maybe you can enlighten us on this, but I suspect it's their very proprietary nature (aka expensive). What do you think?

There's a reason why R2R DAC chips are being phased out, and it's because not only are they expensive but also they reproduce no more details/resolution than the best D-S (and Advance Segment) DAC chips around. Again, unless convinced otherwise, IMHO the culprit on the digital glare being unjustifiably inquisitioned to the D-S DAC chips is the digital filter used that has maximum pre and post-ringing usually used or associated with such D-S DAC chips. You don't think so?

 

You believe that chit? The crux of the issue is the DIGITAL FILTER used, not the DAC chip. The Delta-Sigmas and Advance Segments reproduce the most details and definition from the digital media. What they're saying as lost bits is due to MOST digital filters used, not the D-S or A-S or whatever modern chips.

Again, the real cause of the glare that the R2R fans get their panties in a bunch for is the predominant use of the standard digital filter that has pre-ringing and post-ringing in the higher frequencies. This is the one that's cheap. The standard filter is the CHEAPEST. This is the one that sucks, not the D-S chips. R2R chip using standard digital filter would LIKEWISE sound glary.

The R2R thingy is just marketing to justify the price. Why bash a product to promote oneself is beyond me.

 

And same goes for the 1704 or R2R chips. That is why I think this thread is one great big commercial...the topnotchers were built in part or whole by same people. There's a reason R2R is being phased out...it doesn't have enough definition on bass and treble details. The glare is simply due to the standard (aka cheap) digital filter that has pre and post ringings.

 

 

 

[wmedrz]

It sucks that Purrin is MOT. I really enjoyed reading his opinion about limp d*ck craptastic gears

 

[KeithEmo]

   
44.1/2=22.05. Add in some wiggle room for filtering and whatnot and you get up to about 20kHz (20,000Hz) pretty reliably. Of course, I'd love to see a test where you demonstrate that you can hear differences ±0.1dB at 20kHz.
 
The highest sample rate I've ever seen (for PCM) is 384kHz, and it's really just hardware support. There are like two things recorded in that format. No recorder can use the extra high frequencies. And as Lavry pointed out in that whitepaper, there is a time where higher sample rates introduce more distortion.
 
So, how did you do this 1536kHz test?
 
Also, lossy refers to whether or not a format preserves the original RECORDED signal. So if you can take the original WAV and then take a FLAC you bought off of Bandcamp, invert phase on one of them and get a complete null (silence), you have a lossless format. I don't think anyone would claim that there is a lossless recording method. Recording is a game of compromises and always will be.

 
I've seen a few - very few - commercial 384k files (I even have two of them), but they are VERY few and far between. (I believe I found two websites that each had a few examples of 384k content.)
 
I'm afraid I've got to agree that the white paper about "higher sample rates sometimes causing distortion" is really rather specious. (It's kind of like saying "a TV picture that's too good might make a poor quality projector actually look worse because it struggles to reproduce the unnecessary extra quality".) I cannot fault a content FORMAT for allowing it to reproduce the content more accurately than necessary. If the content producer is worried about 50 kHz noise affecting your playback system, then THEY should limit the bandwidth of their recording - it is not a problem for the recording to have frequency response that's simply better than necessary - unless you're counting on its limitations to protect you from poor production control. (And, if you know that your stereo has problems with a little 25 kHz noise, then it's up to you - or the manufacturer - to "protect it".)
 
A sine wave that is converted into analog using the proper conversion processes and reconstruction filters should be very close to the original. If there were amplitude variations inside the audio band, then they would count as distortion, and so would prevent you from getting the excellent THD measurements that most good quality DACs can deliver. Sometimes you will see amplitude modulations at high frequencies but, as long as they occur outside the audio band, then they are... inaudible. Most DACs do produce "errors" of some sort on transient type signals (which are NOT continuous sine waves), but they tend to be minimal and not all that audibly annoying. (The variations in these "errors" account for the fact that different oversampling filters often sound subtly different, even though they may all have very low steady-state sine wave distortion figures.) However, as has been noted, there isn't a microphone, or a phono cartridge, or a speaker that is able to accurately reproduce transients anyway, so these minor errors are just one of many minor imperfections in the recording and playback chain.

 

[azteca x]

Keith, I'm with you. I'm a big hi-res fan and while I do think 24/96 is plenty it's up to hardware and software makers to make sure that 192 support is just fine. Though I really hope we don't get even bigger file formats. I don't take that Lavry paper as gospel at all, just offering a viewpoint. I have seen HD downloads that are just straight unfiltered DSD, presumably from SACD masters and while I haven't had any issues it is just poor practice. I'm an audio engineer myself and my preference is generally for whatever format it was recorded in to begin with even though sample rate converters are very very good these days ( src.infinitewave.ca ).
 
Anyway, onward.

 

[purrin]

  Depends .... Which one has more taps ?

 
LMAO, I spit my morning coffee on my screen.

 

[goobicii]

20.000hz sine at 96khz,more than double the CD rate,many people say 44.1 is enough and 96khz is ultimate hi rez and than more is stupid

only one cycle out of 5 hit the 100% volume that its supposed,the two lowest cycles are at 86% power,lets say you listen music at 100db peak volume,this means that sine would jump up and down 14db,it would only hit correct volume once in every five cycles...but wait theres more! luckily DACs dont respond instantly and dont follow the samplerate perfectly,they have settling time etc,this gives it rounded edges and actualy makes it closer to sinewave in shape becose this low samplerate draws this sine more in sawtooth shape with sharp edges resulting in distortion..... not only cant 96khz play constant sound to the point its embarassing,it cant even give shape to proper round sine,but but but muh Nyquist but Lavry said... hahaha 

 
 
enter the future,1536khz,perfect amplitude of EVERY single cycle within human hearing range,perfect waveform shape.Oversampling? anti aliasing filters? intersample peaks? lol

 

[Sapientiam]

   
but then you are hearing altered signal not original one

 
What's doing the alteration? The original signal has to be band-limited before it reaches the ADC, so if you're listening with no reconstruction filter on a NOS DAC, its altered from the original.
 
Incidentally you're reading too much into those Audacity waveforms - they're just points joined up with linear interpolation, whereas to get an accurate measure of amplitude you'd need to use sinc interpolation.

 

[KeithEmo]

I'm a little confused by these here pictures.
 
If I sample a 20 Hz sine wave at a 44k sample rate, then each cycle of the wave is represented by more than 2000 sample points - which is going to deliver a pretty darned accurate "connect the dots" drawing. (That top picture looks more like a 10 kHz sine wave, sampled at 44k, then "drawn" without the proper filtering applied.)
 
But what if it turned out that our digital recording did in fact reproduce a 12 kHz sine wave with all sorts of nasty jagged edges, sharp angles, and other distortions - and we didn't filter them out like we're supposed to? What would that sound like?
 
Hmmmmm.... Well, since the second harmonic of 12 kHz is 24 kHz, which is inaudible to human beings, the rattiest looking 12 kHz sine wave imaginable would sound....... exactly like a perfectly clean one. (All we humans could hear would be the 12 kHz primary tone.)
 
However, once we apply the proper reconstruction filter to the output, which filters out the higher frequency junk anyway, what we'll be left with is a 12 kHz sine wave....
 
 
 
 
 
 
Quote:

  20.000hz sine at 96khz,more than double the CD rate,many people say 44.1 is enough and 96khz is ultimate hi rez and than more is stupid

only one cycle out of 5 hit the 100% volume that its supposed,the two lowest cycles are at 86% power,lets say you listen music at 100db peak volume,this means that sine would jump up and down 14db,it would only hit correct volume once in every five cycles...but wait theres more! luckily DACs dont respond instantly and dont follow the samplerate perfectly,they have settling time etc,this gives it rounded edges and actualy makes it closer to sinewave in shape becose this low samplerate draws this sine more in sawtooth shape with sharp edges resulting in distortion..... not only cant 96khz play constant sound to the point its embarassing,it cant even give shape to proper round sine,but but but muh Nyquist but Lavry said... hahaha 

 
 
enter the future,1536khz,perfect amplitude of EVERY single cycle within human hearing range,perfect waveform shape.Oversampling? anti aliasing filters? intersample peaks? lol

 

[Defiant00]

  <snip>

 
As others have pointed out, what Audacity is showing you in your screenshots isn't what a DAC is going to output.
 
You may find this video (with actual examples using an all-analog scope, even) informative: http://xiph.org/video/vid2.shtml