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but I've got plenty of hi res and DSD that I'm tremendously happy with.

 

Me too. DSD includes material from my favorite rock bands (Yes. Genesis, Peter Gabriel, Rush) plus my favorite jazz (on Blue Note).

 

There's far too much for my poor wallet!

 

+1

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Many of the artists you mention are well known for the horrible quality of their recordings. Led Zep, Stones, Nirvana etc. Even the Beatles.

 

But if you like this music, the higher resolution editions offer better sound - and in some cases the best available sound IMHO.

 

Also, there are a TON of great recordings available on both DSD and high res PCM (though to get to the DSD one will need a PS3 that's been configured to rip SACDs). I can't see how someone could today say that this content is now somewhat limited. My want list far exceeds my annual income. :-)

Digital:  Sonore opticalModule > Uptone EtherRegen > Shunyata Sigma Ethernet > Antipodes K30 > Shunyata Omega USB > Gustard X26pro DAC < Mutec REF10 SE120

Amp & Speakers:  Spectral DMA-150mk2 > Aerial 10T

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When hearing differences that can't be measured, the burden of providing evidence falls on those who make the extraordinary claim.

 

 

Some of us prefer to listen to music and not be amateur scientists. I wish they would understand not everyone wants to waste their time doing those tasks. This is why we buy equipment preassembled by good audio designers.

 

+1

"Relax, it's only hi-fi. There's never been a hi-fi emergency." - Roy Hall

"Not everything that can be counted counts, and not everything that counts can be counted." - William Bruce Cameron

 

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Poor Bill Gibson. Utter BS. Really. This is even quite easy to demonstrate. Many have done this before, so I won't go there.

 

For those interested: if you don't see how something like CD can resolve time down to the hundreds of picoseconds, you haven't understood digital audio at all.

 

Well, in my case it is retrospectively obvious, but unless you are thinking about what 16 bit actually means, it is an easy enough mistake to make. (But, then, I wasn't writing an article on Sampling Rates for the AudioPro course).

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But if you like this music, the higher resolution editions offer better sound - and in some cases the best available sound IMHO.

 

Also, there are a TON of great recordings available on both DSD and high res PCM (though to get to the DSD one will need a PS3 that's been configured to rip SACDs). I can't see how someone could today say that this content is now somewhat limited. My want list far exceeds my annual income. :-)

 

Show me where I can find my favourite muziek in hi res. Band like Winter, My Dying Bride, Anathema, Swans, Saint Vitus, Isole, Neurosis, ISIS, Candlemas, Nick Cave and the Bad Seeds, Joy Division, Depeche mode. Or some Classical like Mozarts Requiem by Carl Bohm, or some Karajan recordings?

[br]

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I've spent a lot of thought and experiments on the subject of this thread over the years in my DAC design endeavors.

 

It started when I had built a PCM1704 DAC and listened to it NOS, it sounded very alive, exciting, "pulling you into the music", but "dirty". The infamous aliases were causing "roughness around the edges" to most music.

 

Then I tried a DF1704, which many were touting as the best digital filter to use. Well it was a massive disappointment, the sound was clean, but it was dull, flat, uninvolving, the exact opposite of "pull you into the music". This then started a fairly long quest to find out why the music sounded so dull and lifeless.

 

I won't go into the details of the quest, but the eventual result was that it is the very common practice of cascading multiple filters that seems to be the problem. An FPGA implementing a single SINC filter with the same overall filter characteristics as the DF1704 sounds WAY better. Note that I am not exactly sure WHY the cascaded multiple filters causes this lifeless sound, that has been a much harder nut to crack. The evidence seems to be pointing at the time domain behavior of the filter on transients, but there is a LOT more work to go on fleshing this out.

 

What this has to do with the subject of this thread is that many DAC chips change their upsampling filter depending on the sample rate. At 44.1/48 most use a 3 stage filter, a 2 stage at 88.2/96, a single stage filter at 176.4/192 and no filters at 352.8/384. As far as I can tell this is one of the primary reasons higher sample rates sound better to many people, it has nothing to do with hearing ultrasonics, it's that in many cases going with higher rates cuts down on the number of cascaded filters in the DAC chip, and THAT is what makes it sound better.

 

So if it makes things sound so bad, why do almost all the DAC chips do it that way? As far as I can tell it is a combination of cost and specsmanship. For some reason the stop band attenuation number has become one of the important numbers in a DAC chip spec sheet. (Our chip has 120db attenuations, well OUR chip has 134db etc). Getting these numbers takes a lot of digital horsepower, which would make chips cost too much so the designers need to use things like cascading filters in order to get those numbers for the spec sheets. But if it causes such problems why haven't they come up with other approaches? Probably because the practice is so pervasive that no DAC chip designer has ever heard it any other way. And it doesn't take a megabuck system to hear the difference.

 

BTW this does NOT mean that all single stage filters sound the same. As a matter of fact once you get out of the multistage filter implementations differences in filters can be quite striking. (this is why I think that many people can tell very little difference in the slope settings with most DAC chips that have the options, the affect of the multistage filtering is swamping the differences in slope etc) There is still a lot of work to do in producing really good sounding digital filters.

 

There seems to be a strongly held assumption by a lot of people in this "biz" that the stop band attenuation is the primary metric for a digital filter. This leads to filters with very large numbers of taps, my experimentation seems to point to this being a bad thing. Keeping the convolution kernel small seems to have a much greater impact on sound than the ultimate stop band attenuation. Exactly WHY that is so, who knows, it has to do with human perception and how that wonderfully complicated pattern matching systems takes in the sound filed and determines "that sounds real" or "that sounds not-real".

 

With some decent attention being given to filter design and how it sounds, not just spec sheet numbers, I think the overall level of sound quality from digital audio is going to be increasing significantly in the next few years. Even if the designers don't know exactly WHY it is.

 

John S.

 

Hi John,

 

What are your thoughts and experience (if any) with upsampling and filtering on the PC and then feeding a NOS D/AC?

 

Cheers,

Ric

"Science draws the wave, poetry fills it with water" Teixeira de Pascoaes

 

HQPlayer Desktop / Mac mini → Intona 7054 → RME ADI-2 DAC FS (DSD256)

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Show me where I can find my favourite muziek in hi res. Band like Winter, My Dying Bride, Anathema, Swans, Saint Vitus, Isole, Neurosis, ISIS, Candlemas, Nick Cave and the Bad Seeds, Joy Division, Depeche mode.

 

In your original post on this you wrote that "there is almost zero DSD or High Resolution material available". Had you instead said that "there is almost zero DSD or High Resolution material available for my favorites", I wouldn't have quibbled with you. It's unfortunate that your favorites aren't available for you right now in higher resolutions.

 

Or some Classical like Mozarts Requiem by Carl Bohm

 

Grab this and find someone to rip it for you: SA-CD.net - Mozart: Requiem - Böhm

 

or some Karajan recordings?

 

Some Karajan recordings can be found at High Definition Tape Transfers

Digital:  Sonore opticalModule > Uptone EtherRegen > Shunyata Sigma Ethernet > Antipodes K30 > Shunyata Omega USB > Gustard X26pro DAC < Mutec REF10 SE120

Amp & Speakers:  Spectral DMA-150mk2 > Aerial 10T

Foundation: Stillpoints Ultra, Shunyata Denali v1 and Typhon x1 power conditioners, Shunyata Delta v2 and QSA Lanedri Gamma Revelation and Infinity power cords, QSA Lanedri Gamma Revelation XLR interconnect, Shunyata Sigma Ethernet, MIT Matrix HD 60 speaker cables, GIK bass traps, ASC Isothermal tube traps, Stillpoints Aperture panels, Quadraspire SVT rack, PGGB 256

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Resolution corresponds to 1/2 a period, so 96kHz will resolve the 5.2 microsecond gap.

 

I'm sorry I don't understand what "Resolution corresponds to 1/2 a period" means. I understand the highest audio frequency is 1/2 the sample rate, however Bill Gibson is not talking about the of audio frequencies themselves but how often a sample must be taken in order to capture the fastest level change we can detect. To capture a small dynamic change, especially one unrelated to what comes before and after, as small as 5.208 microseconds wouldn't you need to sample intervals at least 5.208 microseconds apart or 192,000 samples per second?

 

Mr. Gibson used the 5.208 microseconds example to correspond to the common sample frequency of 192KHz. If we wanted 5 microseconds between samples, the shortest level change humans can hear then wouldn't we would need a 250kHz sampling frequency?

I have dementia. I save all my posts in a text file I call Forums.  I do a search in that file to find out what I said or did in the past.

 

I still love music.

 

Teresa

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Hey, mayhem. Disagree with this part of what you said. I'm after simplifying the signal chain - how many digital conversions the file goes through before it gets to the analog stage, for example. And there's no way I'm doing this to be different, separate or elite. I love watching friends and relatives groove on the music, and of course the biggest thing is, I love listening to it myself.

 

Edit: Check out the stuff I recommended to Mordante, especially the Stones if you like them.

 

That part wasn't directed at you.....sorry. And I also agree with the highest resolution recordings.....cause someday....maybe....we'll be able to hear them as intended and beyond.

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I

 

Mr. Gibson used the 5.208 microseconds example to correspond to the common sample frequency of 192KHz. If we wanted 5 microseconds between samples, the shortest level change humans can hear then wouldn't we would need a 250kHz sampling frequency?

 

Sadly Theresa, Mr. Gibson used an incorrect example to support his position.....sensationalist as it was put earlier is very fitting.

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Originally Posted by Jud View Post

 

Hey, mayhem. Disagree with this part of what you said. I'm after simplifying the signal chain - how many digital conversions the file goes through before it gets to the analog stage, for example. And there's no way I'm doing this to be different, separate or elite. I love watching friends and relatives groove on the music, and of course the biggest thing is, I love listening to it myself.

 

 

That part wasn't directed at you.....sorry. And I also agree with the highest resolution recordings.....cause someday....maybe....we'll be able to hear them as intended and beyond.

 

Mayhem 13

I wholeheartedly agree with Jud's statement above.

Get up off your arse and design some speakers that will help to accomplish what you just said.

Feed them using Jud's guidelines, but try and avoid Class D amplification until it's more mature.

There is more than enough crap getting into the mains supply already, and then back into source components.

 

How a Digital Audio file sounds, or a Digital Video file looks, is governed to a large extent by the Power Supply area. All that Identical Checksums gives is the possibility of REGENERATING the file to close to that of the original file.

PROFILE UPDATED 13-11-2020

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I know there are people who believe that high res is snake oil and that anyone who believes they hear a difference, is fooling themselves.

 

In every revolution, you will have laggards.

 

In all my tests, there were benefits with high-res audio. However, I haven't been buying stuff at HDTracks, and therefore most probably haven't had any issues with upconverted Redbook files masqueraded as hi-res audio.

 

That some people still make the supposed benefits or lack thereof revolve around frequency-range reproduction is what makes them lag behind in my opinion. I consider sonic reproduction with a sufficiently high bandwidth a solved problem.

 

A lot of the benefits of hi-res audio are in dynamics, soundstage, timbre accuracy, etc... and, additionally, the better ways you can have at computer player and DAC level to output them (I am thinking here of the better filter implementations for DSD 2x as opposed to some of the bad brick-wall we can see sometimes in CD players).

 

You are right about some of the equipment keeping people from enjoying the benefits.

 

Currently, it takes some work to get digital sounding its best, but this is worth it when you get to that sweet spot (albeit it's a personal one).

 

In the future, with better design, digital will sound close to that sweet spot at un-boxing, with both PCM and native DSD playback optimized so you get the best of both worlds (actually this already exists in the iFi iDSD Nano), non-susceptibility to USB cable differences, etc... so people will not have to worry about all this.

 

The field is nascent but already there has been a lot of progress made very fast. I have noted a catalysis since last year, with nativeDSD, superhirez, Sony getting into the fray.

 

It is not worthwhile to either try to convince the sceptics unless you market related products yourself, nor to worry about how the engineers did their recording/mixing/mastering as you have no control about that but more bits at those stages provide for smoother processing with less artifacts.

 

All that talk about frequency range is a total red-herring... Look for better timbre, wider soundstage (but still realistic), better dynamics when listening to a *well-implemented* digital hi-res system and there should be no doubt at all.

 

All the good things in great analogue systems, but without the wow, flutter, click & pops of vinyl, and the convenience of network sharing of digital, etc...

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That part wasn't directed at you.....sorry. And I also agree with the highest resolution recordings.....cause someday....maybe....we'll be able to hear them as intended and beyond.

 

No doubt there've got to be folks who are into the "My stuff is better than your stuff" aspect, as there are in every hobby and pursuit. Fortunately they're pretty easily avoided.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Teresa, this is in error. Sorry.

 

No apology needed as the science is from Sampling Rates from The AudioPro Home Recording Course by Bill Gibson not from me as I'm not an audio theorist or designer, my interest is the most realism I can capture in the music I love. It sounded logical to me, here is his author bio from Amazon for The AudioPro Home Recording Course Vol. I

 

Bill Gibson, President of Northwest Music and Recording, has spent the last 25 years writing, recording, producing and teaching music and has become well-known for his production, performance and teaching skills. As an instructor at Green River College in Auburn, WA, holding various degrees in composition, arranging, education and recording, he developed a practical and accessible teaching style which provided the basis for what was to come--15 books, a DVD, and a VHS video for MixBooks/ArtistPro along with a dozen online courses for members of ArtistPro.com. Bill's writings are acclaimed for their straightforward and understandable explanations of recording concepts and applications.

 

The sample period is not the limit of time resolution. Indeed perfect redbook could work to portray time differences of as low as 55 picoseconds. Jitter, noise and other factors may increase that to some few hundred picoseconds. But yes we are talking time resolution less than a billionth of a second for most pedestrian DACs.

 

Once again Mr. Gibson is not talking about jitter or time accuracy of the data captured, but how many samples per second are needed to capture the shortest level change that human beings can hear, 5 microseconds which would require 250,000 samples per second to be taken. Test signals are not music. Music is dynamic and level changes of the complex musical waveform between samples cannot be assumed to go in the same direction as they do with test tones. For example, when captured with too low a sample rate then reconstructed back to analog, it will miss or time smear the extremely sharp and loud initial attack of a percussion strike which is many times louder during the first 5 microseconds. This is one of many reasons why recorded music never truly sounds like live acoustic music, although I think 5.6MHz DSD comes close to the feeling of live music, at least for me.

I have dementia. I save all my posts in a text file I call Forums.  I do a search in that file to find out what I said or did in the past.

 

I still love music.

 

Teresa

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I'm sorry I don't understand what "Resolution corresponds to 1/2 a period" means. I understand the highest audio frequency is 1/2 the sample rate, however Bill Gibson is not talking about the of audio frequencies themselves but how often a sample must be taken in order to capture the fastest level change we can detect. To capture a small dynamic change, especially one unrelated to what comes before and after, as small as 5.208 microseconds wouldn't you need to sample intervals at least 5.208 microseconds apart or 192,000 samples per second?

 

Mr. Gibson used the 5.208 microseconds example to correspond to the common sample frequency of 192KHz. If we wanted 5 microseconds between samples, the shortest level change humans can hear then wouldn't we would need a 250kHz sampling frequency?

 

Anyway, I left out a factor of 2^16 in the denominator, so I am off by a few orders of magnitude (see above).

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I'm sorry I don't understand what "Resolution corresponds to 1/2 a period" means. I understand the highest audio frequency is 1/2 the sample rate, however Bill Gibson is not talking about the of audio frequencies themselves but how often a sample must be taken in order to capture the fastest level change we can detect. To capture a small dynamic change, especially one unrelated to what comes before and after, as small as 5.208 microseconds wouldn't you need to sample intervals at least 5.208 microseconds apart or 192,000 samples per second?

 

Mr. Gibson used the 5.208 microseconds example to correspond to the common sample frequency of 192KHz. If we wanted 5 microseconds between samples, the shortest level change humans can hear then wouldn't we would need a 250kHz sampling frequency?

 

Teresa,

 

If you would view this video, starting at the 21 minute mark, you could see a pair of square waves, monitored by a good analog scope show the rising edge can be reconstructed with the edge anywhere in between the actual sample points. Which clearly shows the timing resolution of digital is smaller than the time between samples.

 

Xiph.Org Video Presentations: Digital Show & Tell

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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Think of it this way: Let's say you want to characterize a very sudden impulse, that goes from zero to some large quantity very very quickly (essentially instantly). This is going to look like a step function.

 

If you wanted to resolve the difference between two of these 5 microseconds apart, and you had only one single pure component wave of a single frequency, you would need something fairly high frequency, like 96kHz or 192 kHz (depending upon whether you want to quibble about factors of two). But in reality, you aren't characterizing this with one single solitary component. Even with a CD, you are characterizing it with 65,536 components (which is what 2^16bits gets you). That gives you much more precision.

 

(By "resolve" I mean hear it as two impulses rather than one smeared into the other.)

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Show me where I can find my favourite muziek in hi res. Band like Winter, My Dying Bride, Anathema, Swans, Saint Vitus, Isole, Neurosis, ISIS, Candlemas, Nick Cave and the Bad Seeds, Joy Division, Depeche mode. Or some Classical like Mozarts Requiem by Carl Bohm, or some Karajan recordings?

 

What I've found so far (don't know if these are accessible to you):

 

Search results for 'joy division' | HDtracks - The World's Greatest-Sounding Music Downloads

 

Search results for 'depeche mode' | HDtracks - The World's Greatest-Sounding Music Downloads

 

Search results for 'von karajan' | HDtracks - The World's Greatest-Sounding Music Downloads

 

Vinyl Records, SACDs, DVD Audio, Audiophile Equipment | Acoustic Sounds

 

Edit: Anathema's latest in hi res (the deluxe edition) - https://www.burningshed.com/store/kscope/collection/5/

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Teresa,

 

If you would view this video, starting at the 21 minute mark, you could see a pair of square waves, monitored by a good analog scope show the rising edge can be reconstructed with the edge anywhere in between the actual sample points. Which clearly shows the timing resolution of digital is smaller than the time between samples.

 

Xiph.Org Video Presentations: Digital Show & Tell

 

Test signals are easier to reconstruct unlike complex musical waveforms. I've seen the entire video before and he proved nothing whosoever! His red X over the truth is unacceptable as he used a square wave which has predictable movement compared to actual complex musical waveforms. You've been brainwashed, Monty Montgomery is out to discredit high-end audio and high resolution music. Monty Montgomery is creator of the lossy Vorbis format. Try to understand why people like this tell untruths and misuse test equipment to support their boneheaded conclusions, his is against the interests of all music lovers and audiophiles.

I have dementia. I save all my posts in a text file I call Forums.  I do a search in that file to find out what I said or did in the past.

 

I still love music.

 

Teresa

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