Why is digital audio so complicated? Where did it all go wrong?

[Hifi_swlon]

(Thanks Antoine)

[f1eng]

What makes it all complicated and confusing is discussions like these and the want to be able to fully understand and explain it all. No offense!

Personally I can often only understand part of it, explaining it to others is even harder. While I still would like to understand it all I accept that I, many others and often even manufacturers can't explain in all cases what's going on exactly. We can often at best hypothesize about what's going on. The result is what counts to me.

Thinking we're there and understand it all in our science (which is not the same as engineering) is naive. Again no offense. In a hundred years they'll laugh at our level of scientific understanding, another 100 years later, the same will happen to that generation etc. etc. Scientists often are, in a way, the most stupid (or even corrupt) people on earth, denying/not seeing whatever's right in front of their noses. Creative thinking 'outside the box' is something -many- (so not all) scientists of our age are unable to do, their educations made sure of that. (Unfortunately I've seen it up close/been part of it for some time)

In this hobby I personally like to learn (and thus benefit) from others who've done the painstaking work of experimentation. The good work will always find it's way to the surface, accepted science and even widely adopted engineering principles will no doubt catch up later.

Well I enjoy the hobby too but mainly just listen to music nowadays, I gave up messing around with kit much about 20 years ago after a fairly intense period to find the best sounding kit I could afford and evaluating which bits of kit mattered most by listening extensively at home..

I worked on record players in the mid '70s and nothing new has come up which I did not know then, and  funnily a lot which was well known back then seems to have been forgotten in the new resurgence of interest in LPs.

The basic science on the electrical side of a hifi system is trivial compared to things like radar and microwaves so I am intrigued what you think will be laughed at in the future. Nothing that was going on 40 years ago has turned out to laughable or ridiculous yet - though I suppose we still have 60 years to go.

My oldest recording was made in 1903, so well outside the 100 year ridiculous criterion you espouse, and nothing that they were doing then was ridiculous or laughable, it was using the current materials and knowledge quite ingeniously actually IMO. Yes it was not up to the accuracy of today but it was the best that could be done back then and they were pretty well aware of the shortcomings.

I think the same is still true today.

[SwissBear]

Here is a video related to digital sound processing, distorsions, jitter, aso...
https://www.youtube.com/watch?v=QsVqCID029g. Very instructive in my opinion.

[Krisp]

Krisp suggested it to be error free from source to DAC OUTput. I'm saying it is at best error free up to the D to A conversion. And with error free I mean bit perfect while we're in the digital domain.

The bitstream signal representing the "ones and zero's" is an analog signal. With it travels noise. The signal also distorts while "in travel". The noise and distortion influences the D to A process. Noise, for example in the ground plane, also influences clocks inside the DAC, causing jitter.

...

John Swenson is one of the gifted men who has written plenty or articles on these matters, so I'll just refer to those as a nice starting point. Of course lots more can be found on the net.

http://www.audiostream.com/content/qa-jo...at-digital
http://www.audiostream.com/content/qa-jo...-just-bits
http://www.audiostream.com/content/qa-jo...fect-sound
http://uptoneaudio.com/pages/j-swenson-tech-corner

Sorry I was not precise when I wrote output of the DAC. I meant till the end of the digital path which ends inside the DAC.

And I agree with the noise that is injected especially from the digital curcuits (I read the articles and the postings to the articles), but the noise is just noise (because the modulations of the 1 and 0 are in the megahertz range and the fourier spectrums of those modulations are even higher). All these modulations add up to an irregular random oscillations and the level at which they oscillate can (hopefully) only be heard ( well just the part up until approx. 20kHz or less) when you are close to your speakers and not in your listening position.

As I said, it makes sense to concentrate the "quest" to the analog path after the DAC (if it is a well designed DAC ).


Cheers,

Krisp

[Antoine]

Nowadays source and transport are equally important to me as whatever's behind that DAC and the DAC itself. Of course in the beginning bigger jumps in quality can be made elsewhere but at a certain point you'll reach a plateau. It's a classic discussion; what's more important the source or the speakers. All I know is that those speakers of mine (and I guess of many) are perfectly able to portray even the smaller changes in the area of the source even though those speakers (and in fact all speakers) have much higher distortion levels. They operate on a electromechanical level after all.

I'd like to suggest to those reasoning using textbook knowledge only, to experiment a little. I know an easy one; try to lend a Mutec MC-3+ USB or Intona USB isolator for example, put it in between your source' USB output and the Devialet and report your findings back here. I'll guarantee you: same bits, more music!

OK, with all due respect, now I am really retiring from this topic. Yes I know I returned by own choice but I already regret it. There's nothing in it for me personally to convince nay sayers. In fact it's worse; it's an energy drain for me. People can all believe what they want to believe, please don't mind if I'll do just the same.

[Jean-Marie]

The most complicated and less mastered element in the whole digital chain is our brain....

Jean-Marie

[f1eng]

The Mutec MC-3+ USB can reduce jitter on a SPDIF source, as long as the source has a less good clock than the Mutec, and may improve a USB source if the clock in the receiver is less good than the Mutec.
This is straightforward and well known for decades digital technique, though the product is new. I bought one after reading about it here.
I have not done a rigorous compare with it yet, so I don't know either whether the Mutec clock + digital connection is lower jitter than the clock in the Devialet right next to the DAC or whether the jitter threshold is below audibility already if it is.

In the case of my digital recorder where the output of the ADC/DAC is audibly indistinguishable from the microphone feed, the ADC and DAC share a clock, pcb etc. and the signals are internal and short path, so best case scenario, but there are no exotic parts and no fancy box so it is less expensive than most audiophile stuff and is, I can assure everybody, completely transparent for the type of music (classical) I listen to and record (to my ears) and that has rarely been the case.

[yabaVR]

Very interesting thoughts so far…..

 ... I would love to take part i something like that, to see if a raspberry Pi DAC/Amp versus a Naim Statement or whatever could be picked out blind, I find it fascinating.  Anyway, drifted off my own thread!

+1
But I would like to put the system together by myself. I think the problem with double blinds is that they are done by people that are expecting to hear no differences. So they lack of skills to put the system together the right way. No offence, but let it be done by someone who knows the gears. For example I would give full trust to the experience/knowledge of Antoine to put together such a sytem.
Let the nobelievers do the whole administrative procedures, I'm ok with that.

If you got a perfectly tuned system playing at top level there is no chance you prefer a mp3-file against a 16/44.1, even if the "normal people testers" degenerated to only listen to mp3 nowadays. 

So these popular double blinds on the net are a nonsense and only feed the nobelievers expectations. They are like your local doctor doing a heart transplantation. Sure he's a doctor, but he will/has to fail (this includes some nobelieving pros, recording studio engineers, etc.).

This field is a very wide one and it has to be worked out by experts ("heart specialists") and even the experts run into occasions they have a hard job to deal with.
It's no black and white at all and we can fall into traps just by chance. I see that every day.

gui

[yabaVR]

So, I've been thinking about this quite a bit over the past months, following topics all over the place on various forums etc - including here obviously -  as well as from my own experimentation.

Lets just get this out there first - I'm not sure how factually correct any of this, it just one persons opinion - but as far as I can tell, CD was invented about 30 years ago(?) and it bought digital audio to the masses, but many still argue to this day that it takes an ultra-expensive player to make CD sound 'good', and that's still miles from being as good as vinyl, or at least from sounding 'realistic' or 'pleasing' (read natural I suppose).  Computer based audio followed, and it seems like only an uber expensive computer based setup can sound as good as an uber expensive CD player, and in most cases the sounds is still only just believable as being 'real' and 'natural' in many cases, although some say high-end digital is now equal or better than vinyl, so it seems it can be done.

In my mind, the theory's simple - digital data is sent from A and received at B, where it is converted to analogue for output.  If dropouts aren't occurring, then we assume the data arrived unharmed, but that's not quite good enough apparently.  (OK, there are a lot of things in the chain like DACs, amps and speakers but lets ignore those for now and assume they're perfect.).  The bit I'm talking about is this getting the digital data from source to destination, and the related problems that are bounded around.  Jitter, EM interference, Signal Integrity, clock bending, power supply noise, ground noise, galvanic isolation, atomic clocking, digital cable boundary effects, cable impedance, reflections, the list goes on I'm sure - but all seem to contribute in some way towards making this specific audio version of digital, somehow prone to lots of issues, and there is no length you can't go to extract a bit more just by making this digital data 'better'.

Now, we've been to the moon, and are planning to go to Mars, we're starting to get a much deeper understanding of the universe and how it was formed, we've made radical advances in medicine and science - many of us carry computers in our pockets that are more powerful than the early supercomputers.  There's a whole swathe of other things too many to mention here obviously.  (OK so we're not so smart because we've also f'ed up a lot of the planet and caused untold suffering at the same time…but technology-wise we seem to know what we're doing).

So what went wrong with digital audio?  Why has something that in this day and age should be so easy, turned out to be so complicated?  Are the problems all real, or  part real, or is the actual problem just not really understood? Or are there a lot of people taking advantage of the fact that its hard for humans to make decisions about what they hear so are deliberately misinforming or making it more complicated than it needs to be? Or a combination of?

I've done a fair bit of tweaking - albeit nothing compared to some - and think I'm open-minded, but often at the end of it I start to question myself, and wonder whether those that say it's just the brain playing tricks might be right (at least in part).

Be keen to hear what others think….

Back on topic.

I think that the process and handling of digital data is kind of more difficult than the inventors thought. I mean, it's like having the tools to cut a banana into pieces and glue it together again...we know we have the ability to do that...but it nevermore is the same banana again. We just don't have the skills to get back to its origin (until we are on the USS Enterprice and use the replicator ).

There is so much time-phase sensitive processes in digital audio data, so the ADC/DAC can not glue the pieces to the right place in time beside loosing some pieces (CDP) and interpolate them.

If you use a complete analog system you have only minor problems with time-phase. There is a continuous music signal and it's not cut. I think that's the first obvious you get when you are listening to analog music...and due to lack of words you call it naturalness. All the rest like distortion and frequency characteristics are not that much disturbing your ears/brain like that one with the time-phase. For me it's the major reason why so many swear that LP sounds best for them and I have the same quaint feelings when I listen to it.

One other thing is that a digital signal in its origin still is an analog signal. And this is made of frequencies and electrons and...
There are so many things that are carried with the digital signal. It's by far not only the 1s and 0s. Sure you can filter some of the junk but all of it?

Hard core scientist even discuss the possibility that an electron could have a memory. So there you go...if an electron not only having a charge/spin but also some other characteristics you can merely imagine on the consequences for your audio signal being made of it.
So far we come, the less we know   .

gui

[f1eng]

So, I've been thinking about this quite a bit over the past months, following topics all over the place on various forums etc - including here obviously -  as well as from my own experimentation.

Lets just get this out there first - I'm not sure how factually correct any of this, it just one persons opinion - but as far as I can tell, CD was invented about 30 years ago(?) and it bought digital audio to the masses, but many still argue to this day that it takes an ultra-expensive player to make CD sound 'good', and that's still miles from being as good as vinyl, or at least from sounding 'realistic' or 'pleasing' (read natural I suppose).  Computer based audio followed, and it seems like only an uber expensive computer based setup can sound as good as an uber expensive CD player, and in most cases the sounds is still only just believable as being 'real' and 'natural' in many cases, although some say high-end digital is now equal or better than vinyl, so it seems it can be done.

In my mind, the theory's simple - digital data is sent from A and received at B, where it is converted to analogue for output.  If dropouts aren't occurring, then we assume the data arrived unharmed, but that's not quite good enough apparently.  (OK, there are a lot of things in the chain like DACs, amps and speakers but lets ignore those for now and assume they're perfect.).  The bit I'm talking about is this getting the digital data from source to destination, and the related problems that are bounded around.  Jitter, EM interference, Signal Integrity, clock bending, power supply noise, ground noise, galvanic isolation, atomic clocking, digital cable boundary effects, cable impedance, reflections, the list goes on I'm sure - but all seem to contribute in some way towards making this specific audio version of digital, somehow prone to lots of issues, and there is no length you can't go to extract a bit more just by making this digital data 'better'.

Now, we've been to the moon, and are planning to go to Mars, we're starting to get a much deeper understanding of the universe and how it was formed, we've made radical advances in medicine and science - many of us carry computers in our pockets that are more powerful than the early supercomputers.  There's a whole swathe of other things too many to mention here obviously.  (OK so we're not so smart because we've also f'ed up a lot of the planet and caused untold suffering at the same time…but technology-wise we seem to know what we're doing).

So what went wrong with digital audio?  Why has something that in this day and age should be so easy, turned out to be so complicated?  Are the problems all real, or  part real, or is the actual problem just not really understood? Or are there a lot of people taking advantage of the fact that its hard for humans to make decisions about what they hear so are deliberately misinforming or making it more complicated than it needs to be? Or a combination of?

I've done a fair bit of tweaking - albeit nothing compared to some - and think I'm open-minded, but often at the end of it I start to question myself, and wonder whether those that say it's just the brain playing tricks might be right (at least in part).

Be keen to hear what others think….

Back on topic.

I think that the process and handling of digital data is kind of more difficult than the inventors thought. I mean, it's like having the tools to cut a banana into pieces and glue it together again...we know we have the ability to do that...but it nevermore is the same banana again. We just don't have the skills to get back to its origin (until we are on the USS Enterprice and use the replicator ).

There is so much time-phase sensitive processes in digital audio data, so the ADC/DAC can not glue the pieces to the right place in time beside loosing some pieces (CDP) and interpolate them.

If you use a complete analog system you have only minor problems with time-phase. There is a continuous music signal and it's not cut. I think that's the first obvious you get when you are listening to analog music...and due to lack of words you call it naturalness. All the rest like distortion and frequency characteristics are not that much disturbing your ears/brain like that one with the time-phase. For me it's the major reason why so many swear that LP sounds best for them and I have the same quaint feelings when I listen to it.

One other thing is that a digital signal in its origin still is an analog signal. And this is made of frequencies and electrons and...
There are so many things that are carried with the digital signal. It's by far not only the 1s and 0s. Sure you can filter some of the junk but all of it?

Hard core scientist even discuss the possibility that an electron could have a memory. So there you go...if an electron not only having a charge/spin but also some other characteristics you can merely imagine on the consequences for your audio signal being made of it.
So far we come, the less we know   .

gui

Oh dear Gui, what a shame the internet has no peer review or knowledgeable editor.

What you write here has been, more or less, repeated many times on the internet forums but repeating something which is not true over and over again will not make it become true, however plausible the erroneous statement may appear to be to those who do not understand.
The Nyquist–Shannon sampling theorem is not a theory, it is mathematically proved.

There are plenty of reasons why digital audio may not be to somebodies liking and there are plenty of ways in which it can be implemented imperfectly, but what you write is completely imagined gobbledygook. 

I am not religious either nor do I believe that homeopathy or acupuncture are anything other than placebos. Blind faith and belief in the implausible is not my thing, so excuse my being blunt.

Some of this is my opinion, some my experience of 50 years recording and some is straightforward mathematics (I concede that very few people have any mathematical aptitude though)

And don't think I have not listened, I am very sceptical and have always believed both that if theory and practice differ that the theory is incomplete and that if 2 things measure the same but sound different the wrong thing is being measured. I always check everything from first principles myself and have done many listening evaluations over the last 45 years.