Worlds Greatest DAC and what it does differently

[beerandmusic]

 

First off I am a layman when it comes to audio engineering....i know very little, even less than 10 percentile of those active on this forum. I will have ridiculously stupid questions, and will want to understand things on the most simplest terms. I am ADD and a bit scatter-brained, so please work with me, as many times I may think that what others believe are not related where I do. Anyway, on to the topic at hand.

 

Of course the greatest DAC in the world will do many things more that my "hypothetical" dac will do...but for the purpose of this discussion, there is only one thing this DAC will do that no other DAC can do..

 

In my understanding the purpose of the DAC is to "accurately" recreate the analog waveform that which was recorded from analog to digital.

 

To keep things "simple" the only purpose of this dac is to output a near perfect 1khz sine wave that was accurately recorded and digitized into a 1 second 44.1K wav file.

 

IMHO, The person that is deserving of the most respect on this webiste (MISKA), suggests not only can a DAC not accurately reproduce a 1K sine output, but that the final analog output that is just trying to accurately recreate this 1K sine wave, will be "VASTLY DIFFERENT" between 2 dacs, say a Schiit MB PCM dac and an RME ADI-2, even given all the other same hardware and source file.

 

If DAC engineers cannot create a DAC that I would consider (as a layman) should be a relatively simple chore, then we need a DAC that is capable of doing this FIRST before we worry about more advanced features.   Is it not possible for a DAC engineer (or even a team of the most skilled engineers) to create a DAC that is capable of reproducing a 1K sine wave that is relatively accurate?

 

Also, if modern day DACS are not even capable of performing with some bit of accuracy the most simplest of functions, then imho it would be ignorant to pay much for any dac.

 

 

[beerandmusic]

20 minutes ago, beerandmusic said:

 

First off I am a layman when it comes to audio engineering....i know very little, even less than 10 percentile of those active on this forum. I will have ridiculously stupid questions, and will want to understand things on the most simplest terms. I am ADD and a bit scatter-brained, so please work with me, as many times I may think that what others believe are not related where I do. Anyway, on to the topic at hand.

 

Of course the greatest DAC in the world will do many things more that my "hypothetical" dac will do...but for the purpose of this discussion, there is only one thing this DAC will do that no other DAC can do..

 

In my understanding the purpose of the DAC is to "accurately" recreate the analog waveform that which was recorded from analog to digital.

 

To keep things "simple" the only purpose of this dac is to output a near perfect 1khz sine wave that was accurately recorded and digitized into a 1 second 44.1K wav file.

 

IMHO, The person that is deserving of the most respect on this webiste (MISKA), suggests not only can a DAC not accurately reproduce a 1K sine output, but that the final analog output that is just trying to accurately recreate this 1K sine wave, will be "VASTLY DIFFERENT" between 2 dacs, say a Schiit MB PCM dac and an RME ADI-2. 

 

If DAC engineers cannot create a DAC that I would consider (as a layman) should be a relatively simple chore, then we need a DAC that is capable of doing this FIRST before we worry about more advanced features. 

 

 

 

And just for reference purposes, does anyone know what a 1khz signal would look like digitally?

say 1/10 of one second recorded at 44.1K and assume "cleanest" with no attenuation....or in layman's terms, the "most simple".

[beerandmusic]

9 minutes ago, pkane2001 said:

 

If you are looking at the differences in the human-audible range, these "VAST" differences would be a VAST exaggeration. A 1kHz sine wave reproduced by most competently designed DACs (possibly excluding some NOS R2R types) will look like a 1kHz analog sine wave. You'll need to zoom in with a magnification of 10,000x times or more to start to see some minor differences, and that would be with some of the poorer performing DACs.

 

Thank you sir!  I have always known that you are well respected for your knowledge but have been a bit of a pain in the past, so this type of response to a layman such as myself that is just seeking to gain knowledge to make a knowledgeable purchase is well received and appreciated.

 

So in your opinion, when I asked MISKA what are differences in the audible range, he is exaggerating to say they are VAST.  That is what i believed to be true.  I understand everyone has a product they want to sell or to be on one side of a camp or another...i am just trying to make an educated decision based on my desires without any bias or desire to create conflict...so thanks again....let me gather my thoughts based on that response for my next question....

 

I really am trying to get to a certain point, but it will likely take me a few days to get there based on how many accurate responses i get.

 

I guess my next question would be do you consider Mike Moffat's (schiit) current DACs a competent design.

It is on my shortlist.

[beerandmusic]

16 minutes ago, pkane2001 said:

 

Compliment received and acknowledged 😜

 

To be honest, i have always believed you would probably be the best person to answer my questions, as I know Miska is both biased and busy, but did expect relatively accurate answers from him (although for some reason he seems to purposely "beat around the bush").  In the past, I have been reluctant to even ask you a question, but right now at the moment, I am feeling differently (smile).  I know i can be antagonizing, and generally disliked by many in my manner of speaking, even though I have no idea why...(It is likely my ADD and passionate quest for logical reasoning) but it has been my case my entire life...i am just me.  I retired young mainly because of my inability to be socially acceptable and lack of sleep.

 

Anyway...i edited the question above asking about Schiit...I have owned his stuff before and have liked it, but in the past i just wanted DSD, because i "think" in my mind, it was better, but ever since i got rid of the schiit, i have thought it had some "magic juice" that other dsd dacs have not...and I think i have gained "a small amount of knowledge" where i think I am more acceptable of PCM now ....anyway, that will come later...there is no reason to go further down where I am trying to go, without first a response to my question above (in previous post that i edited), about Schiit DACS.

[beerandmusic]

Before we can get where I am hoping to get, there has to be a consensus on some very basic principles (at least to me).

 

The first phase of my objective is to see if we can get a concensus, that whether a 1K sine wave recorded at 44.1k will "sound the same" whether it is played natively, upsampled to 192K PCM or upsampled to quad DSD.

 

Initial Assumptions (will modify these later):

.wav file in it's most simplest form

same quiet USB source

Competent well designed DAC

all other hardware the same (assume competent mid-fi)

simple chain using USB compliant cable and standard 16gauge copper speaker wire)

 

 

[beerandmusic]

1 hour ago, fas42 said:

 worrying about a 1kHz waveform is pointless,

 

This effort has nothing to do with music in itself, it has do with a project of mine to decide if i want to consider a PCM dac.

My current belief is that I can be very content with PCM, whereas I used to believe that i would not ever purchase anything that did not support DSD...so you just have to have patience to see where I am going.  My guess (but i may be wrong) is that most audio engineers already accept PCM is more than sufficient and capable of playing the full audio spectrum equal to quad DSD "if done properly".  I still do not understand even the smallest fraction of what most here do, but i "think" i know more than I did "yesterday".  Anyway, to me, and I started the thread, it is not pointless...it is a starting point for a debate to see if there can be a consensus.  If there can't be a starting point, there is no reason to dig deeper.

[beerandmusic]

39 minutes ago, asdf1000 said:

 

Can you share a link to this post of @Miska ?

 

I don't know if he misunderstood my question or , but i laid out clear assumptions::

1.  All hardware (besides DAC) is the same

2.  Source File is the same (a wav file recorded accurately at 44.1k)

3.  Assuming no DSP EQ or any other changes to original file besides upsampling.

 

Wanting to know if the original recording was 44.1k, is it possible if a 192K PCM DAC is capable of sounding the same as a QUAD RATE DSD upsampling of same file, i wanted to know what the measurable differences within the audible spectrum of a DACs output (at analog out) might be between the 2 different DACS....keeping in mind that the original recording was done at 44.1K and that the DACS only purpose (for this exercise) is to accurately convert the digital signal back to analog that was first recorded.

 

When he said they would be massively different, it kind of caught me by surprise, and so i questioned further....what if the original recording was just a 1 second 1KHZ sine wave, and he repeated that the differences in the audible range would still be VAST/massively different.  And i thought, if the main purpose of a DAC is to accurately reproduce the digital signal that was first recorded, and it can't reproduce a siimple 1khz sine wave with relative accuracy, there is something seriously wrong with audio engineering, and why spend crazy amounts of money or chase something that can't be caught.

 

 

Anyway, i am sure there can be a starting point with a broad consensus to start, but this effort may take awhile.  I know DSD vs PCM has been argued many times before, but I think for me personally, i have a "slightly" better understanding now than I did last time, so wish to re-hash it.

 

 

[beerandmusic]

17 minutes ago, asdf1000 said:

 

He’s not talking about what differences can and can’t be heard.

 

Yes, he was...i specifically stated in the thread, "differences within the audible spectrum" and i repeated it many times.

[beerandmusic]

2 minutes ago, asdf1000 said:

There’s nothing controversial about this.

 

The controversy is all in what differences can and can’t be heard but I’m not going there. That topic has been bashed on every audio forum for years.

 

I have pieced together many quotes from many respected people, so i still have questions....

Just curious, what is your take on if PCM 192K can sound the same as QUAD RATE DSD provided original source is 44.1K?

[beerandmusic]

3 hours ago, yamamoto2002 said:

 

When 1 second of 1kHz signal is sampled by 44100Hz, 44100 sampled values is produced. (This is obvious🙂 )

 

When we look closer to those 44100 samples, first 441 sample values are repeated 100 times. This is caused by periodicity of sine function : sin(x + 2π) = sin(x). This 441 samples contain exact 10 cycles of sine wave.

 

Most of sampled values are irrational numbers.

When those values are quantized to 16bit or 24bit integer PCM, sample values becomes something like the following table.

 

In digital domain, PCM signal is stored/transferred as a list of those integer values. One second of PCM consists of 44100 integer values.

 

First part of the sampled values plotted using Audacity:

 

DAC output analog waveform simulated by Adobe Audition:

 

 

 

Thank you sir!  That is what I love about this site.  There is so much knowledge people are willing to take their time and share.  Do you sell any products?  You don't write many posts, but when you do, I am sure it can be trusted.   I noticed 2 things about you but wish I knew more.  1. You don't have an "about me" in your profile...please do edit.  2. Your system is a modest Marantz and B&W system....a man after my own heart!  Anyway, if you wish to remain "annonymous" I understand, but I would love to know more about you.

[beerandmusic]

4 hours ago, yamamoto2002 said:

 

When 1 second of 1kHz signal is sampled by 44100Hz, 44100 sampled values is produced. (This is obvious🙂 )

 

When we look closer to those 44100 samples, first 441 sample values are repeated 100 times. This is caused by periodicity of sine function : sin(x + 2π) = sin(x). This 441 samples contain exact 10 cycles of sine wave.

 

Most of sampled values are irrational numbers.

When those values are quantized to 16bit or 24bit integer PCM, sample values becomes something like the following table.

 

In digital domain, PCM signal is stored/transferred as a list of those integer values. One second of PCM consists of 44100 integer values.

 

First part of the sampled values plotted using Audacity:

 

DAC output analog waveform simulated by Adobe Audition:

 

 

 

The reason, i wanted to know about a "simple" 1K signal is because in the audio world it is simple, finite, and predictable.  I believe in taking a system down to it's bare minimum for troubleshooting purposes.  If there is a problem in a bare minimum configuration, obviously that would need to be addressed first.  If there is "any" possibility that there would be any audible differences between 20 different dacs at any price point, then those should be discounted.  Everyone suggests different DACS sound differently.  This is a real problem and tells me that audio engineering is nowhere near where it needs to be.  The function of a DAC is to "ACCURATELY" reproduce a finite signal that has previously been accurately converted to digital.  DACS should not sound differently.  You shouldn't have to "pay" for what sounds better, as if a DAC is doing its job properly, all DACS should sound exactly alike (given same source and other hardware).  They shouldn't color to meet our emotions....anyway, I am sure you are well aware of what I am saying and thanks for sharing...

[beerandmusic]

I noticed a few more have responded.  Thank you. 

I will take the time to read through carefully.  Again, I am a layman, and most of this stuff is way over my head.

I am "extremely" left brained, which is likely why I am socially dysfunctional.  My logic differs than most peoples.

I am also ADD...which is why I am going to go here now:::

 

Someone commented in another posting, that I made a rather offensive remark, where in my thinking, it was not offensive at all.  Quite the opposite.  In my obscure way of thinking, honesty conquers all other forms of communication, and anything that is less than honest is a "disservice" to humanity.  I believe everyone should be "candid" with their true feelings and emotions.  I believe that if everyone did this, that at first there would be a lot of chaos and death, but once it stabilized, and universally accepted, there would be more harmony, understanding and acceptance.  I am sure someone will give me an "off topic" for these thoughts, and that is fine.  This is my thread, and where someone may feel it is off topic, I do not.  It tells you a little bit about my character, and where I am going with this thread.  I do not seek conflict, I am not biased, and I only seek understanding and harmony.  Of course, I realize it will be a rocky road getting where I am going.

 

Anyway, enough of my ADD ...switching gears....I will read through carefully, and respond soon, in my layman's manner.  Time for lunch. 

 

What i currently believe, but am open to anything, is that whereas I do believe the source matters, that the better the source is, the less upsampling matters....and the most important part of this adventure, will to be keeping in mind, that if a DAC is doing it's job properly, then all DACS should sound the exact same knowing they have a finite input of a previously recorded signal and their only function is to be accurate in recreating that signal.

 

Edit to add::  knowing the original signal has been digitized, engineers should be able to know if their DAC is functioning properly by comparing the input signal (prior to being digitized) to the output signal.  If there is any differences between dacs, then at least one of them is faulty (keep that in mind).  When troubleshooting a DAC design, engineers should compare the input to the output gradually adding more things in to see when there is a difference on the output.  Initially, testing simple analog signals, and then continue to be more complex. 

 

Lunch time...

 

 

[beerandmusic]

23 minutes ago, barrows said:

In a "perfect" world, where electronics all acted "perfectly" with no errors or losses' and where digital filters were "perfect", with infinite stop band attenuation and no artifacts, then 16 bit 44.1 kHz sample rate would result in perfect output from a DAC.

 

But nothing in the real world works "perfectly", everything is subject to errors in one way or another:  digital filters produce artifacts (alias products and or ringing), analog circuitry is subject to interferences from high speed processor noises which may produce intermodulation effects in the audible bandwidth.  Electronics are subject to thermal noise at very least, and harmonic distortions.  DACs are mixed digital and analog circuitry in a single chassis, which in itself is a challenge.

 

Higher sample rates, and oversampling overcome some of the real world limitations of how things actually operate.

 

I haven't read through yours or davides posts yet, so i will get to them later, but this posting captured my attention.

I have lived in the digital world for my entire career and things are more "predictable", so maybe this is where my logic fails in the audio world....but i want to concentrate on where things break down....i think most people would agree that most dacs would produce the same output for a simple sine wave, so it is a good starting point....anyway...thanks for your input as always...will continue to digest.

 

edit to add:: since you believe if everything was perfect that 44.1khz should be sufficient to capture the audible spectrum, I personally think that 96K would likely be more ideal as 44.1K is very close to our boundaries and that 96K "may" be a rudimentary step that would leave no room for argument, and plus it is a multiple of 48k which I understand is more ideal for video, that 96K "may" be a "sweet spot" for both recording and reconstruction.

[beerandmusic]

I may pop in from time to time simply to post a thought that entered my mind, that may or may not be applicable, but at the moment I post it, I believe it is applicable, and don't want to forget the thought.....

 

this is such a moment, and just want to share this thought....

 

Part of the problem with DAC design may be the amount of different inputs it has to resolve for, that if the design could concentrate on a limited amount of inputs, the design could be better perfected....along this thought, if 44.1K fully encapsulates our audible spectrum, that if the DAC only had to resolve for example 44.1, 48, and perhaps 96k, it may be easier to perfect the design to accurately reproduce the input signal, if it had less variables to deal with.

[beerandmusic]

another thought...i think bit depth is important for dynamics, and usually when i do critical listening, i look for dynamics in very quiet background sounds, and is where i typically find differences.  I understand there is a theorem (nyquist) for sample rate and this is probably a very stupid question that i could probably google to find the answer, but want to continue to demonstrate that I am a layman, and that no question is a stupid question, so..the question is, does this theorem take into consideration bit depth?

 

again, this is just a rambling thought that crossed my mind, that i do not wish to forget, so just posting it here for future consideration....

[beerandmusic]

^^^  I think that once the "end to end" D->A design is perfected, and mass produced, that it could be manufactured based on BOM for very little money....of course, initial runs, to be profitable and to carry engineering costs would be astronomical...but a lot of knowledge is already known...

My thinking is that since we have a finite digital signal, the entire analog signal could be accurately mathematically reconstructed via software (and i am not sure i understand any need of high rate dsd to reconstruct a 44.1k recording), and that the actual analog out really wouldn't need expensive power or isolation circuitry....but this is a different topic, and just a concept in my mind at the moment...I would have to hash the thoughts around a lot, before i could say this confidently....but currently i have much lower level understanding needed before i get to that point.

 

 

 

 

[beerandmusic]

45 minutes ago, barrows said:

The reason for the DSD 256 conversion rate is because this makes the actual conversion process much much simpler and more accurate.  Resulting in improved sound quality.

And who said we should start with 16/44.1 music recordings, those are compromised to begin with.  I will choose a higher rate file any time I have the option to purchase it, because there is an advantage to higher rate of recording.

 

BTW, right now I am listening to a DSD 128 file of Debussy Piano works recorded by Wave Kinetics records, played by Ilya Itin, through the native single bit discrete conversion of the Bricasti M3 DAC.  It sounds AMAZING!

 

I agree, i don't think we should start with 16/44.1 but that is what mostly exists...but my current understanding is there likely wouldn't be any benefit to higher than 96K sampling rate( and unsure of if there is a theorem representing optimal bit depth?).

 

I have many native DSD256 and DSD128 files and have downampled them out of curiosity and couldn't hear differences, whereas a quieter source and better bit depth have made an audible difference. I also agree, It may make no sense to downsample if we have the technology to "ACCURATELY" record and reproduce at higher rates, then why not (other than wasted power).  My issue is that there are bigger issues to resolve for than potentially unnecessary higher sampling rates. 

 

If i play a 320k mp3 file from a good player, it will sound a lot better than a DSD256 file playing from a noisy local pc.  We need to identify why we hear differences in different dacs and why the source matters so much.

 

I really am kind of spacing out right now, and not sure where I am trying to go with this entire thread, as I have new discoveries since i started it (grin).  I guess I believe that a well designed DAC even with a "lowly" 96K should be as capable as a quad dsd dac, and you have already surrendered that in a perfect world a 44.1K dac would be sufficient.

 

I am curious as to what is the threshold of the break down is....if you believe a 44.1K should be able to be 100% accurate in reproductions of a 44.1k recording. There has to be a point where one can confidently say that under certain "defined conditions" that a 44.1K dac will be perfect.... again, I don't even like using 44.1K as i believe in "golden ears" and room for correction, and why not 96K....i am only using 44.1k for this hypothetical dac because that is cd rate and what most recordings were created and it is of lowest denomination as a "starting" point....an am just as happy to just start with 96K since so far it seems there is a concensus that every dac can reproduce a 1Khz test signal with relatively 100% accuracy.

[beerandmusic]

REFOCUS::  We can discuss higher recordings later...I think for now i want to stick to resolving for original recordings at 44.1K still...I personally am not ready to move on from that at this time for my purposes....i think it is agreed, by at least many, that original recordings recorded at 96K with higher bit depths, may have benefits...... but most of what is in libraries today is 44.1k....so sticking with that "for now".

 

On that assumption, i believe most people would agree it is "possible" to accurately reproduce the original signal with a 96K sample rate.  The downside to upsampling higher is the possibility that better clocks, better power, and more advanced filtering is needed, which in of itself could add need for more money and wasted power if it is not necessary to "accurately" reproduce an original recording done at 44.1K.

[beerandmusic]

7 hours ago, yamamoto2002 said:

 

When 1 second of 1kHz signal is sampled by 44100Hz, 44100 sampled values is produced. (This is obvious🙂 )

 

When we look closer to those 44100 samples, first 441 sample values are repeated 100 times. This is caused by periodicity of sine function : sin(x + 2π) = sin(x). This 441 samples contain exact 10 cycles of sine wave.

 

Most of sampled values are irrational numbers.

When those values are quantized to 16bit or 24bit integer PCM, sample values becomes something like the following table.

 

In digital domain, PCM signal is stored/transferred as a list of those integer values. One second of PCM consists of 44100 integer values.

 

 

amazing that much data is used in a 1 second sampling of a simple 1k sine wave.

I am confused as how does this data look digitally in a flat file?

[beerandmusic]

8 minutes ago, yamamoto2002 said:

 

There is actually infinite there in the discrete-time sampled 1kHz sine signal:

• Most of sample values are irrational numbers. This means, to represent one sample value exactly (for example sample#148 = sqrt(3)/2 ), infinite number of digits are needed. 16bit quantization truncates the value of first 16bit of the number to create finite digit rational number and quantization noise is generated by the truncation.
• Pure 1kHz signal is, by definition, has infinite length. If it is truncated to finite length (1 second), other frequency components appear on the truncated edge and signal is contaminated. You may hear click noise at the truncated edge.

thanks.  I guess i should have said the digitized flat file would be more simple and easier to compare the original analog input to the analog output (assuming differences would be easier to be measured?)  There may be an easier test for accuracy, but as a layman, that just seemed like a good example as a starting point.

[beerandmusic]

29 minutes ago, yamamoto2002 said:

 

I choose WAV file as an example to explain it.

 

WAV file stored 1 second (truncated) 1kHz sine can be created using WaveGene.
Generated WAV file can be read using any Hex editor. I'm using HxD.

 

 

 

 

thanks, i realized it would have a header, and assume an acknowlege of some type (even for isosynchronus?), it appears what is actually recorded is the offset rather than a value, and it does appear linear.  What does the first negative number represent, and will it eventually repeat?

I just noticed your tagline Developer of PlayPcmWin...will have to check it out.

 

Edit to add:: I just downloaded your player and it sounds great.

 

 

[beerandmusic]

13 hours ago, davide256 said:

I have found that DSD upsampling matters with CD quality choral pieces; there is normally a fuzziness associated with massed voices that up sampling removes, allows inner

counterpoint singing to be heard clearly vs covered up. I've heard this also with better PCM gear but the price tag becomes unaffordably expensive.

 

 

what is an example cd i can test what you are speaking about?  Can you share a cd, track and time that you are talking about? 

[beerandmusic]

2 hours ago, yamamoto2002 said:

 

1kHz 1 second 44100Hz PCM starts from 441 unique sample values and it is repeated 100 times. Please refer my first post of this thread what it is like this 441 sample values are.

 

ok, thanks now it makes a lot more sense to me....only "value" columns are the actual data in the flat file, and you are showing it expressed in hex vs binary.  As simple as the signal is, it still requires a lot of unique bits (smile)....still it should be able to be converted to an analog wave form out with relative precision, and should the same no matter what dac was used or how it is upsampled.

[beerandmusic]

 

For my purposes, the fact that Barrows conceded that with "perfect" electronics, you can't better sound than what a 44.1K PCM dac would do, and that no upsampling would give you better sound....I was even willing to concede a 96K PCM dac would do....(and actually is closer to what my belief is).

 

That acknowledged, that also allows for my statement "the better the source" the less upsampling will make any difference.

 

I never saw an answer regarding if there is a theorem for what bit depth is required for human hearing, but i found this

https://www.soundguys.com/audio-bit-depth-explained-23706/

so if 20bits is the magic number, i will just go with 24 to be safe.

 

I guess to surmise, then it would be accurate to say the worlds greatest dac, will compensate for imperfect electronics, and present an accurate reproduction of the of the original analog recording, and does not need to be any more sophisticated than a 24 bit 96K PCM DAC.

 

Not sure I need to go any further with this topic...but it may raise a new topic (smile).

[beerandmusic]

^^^^ Thinking more along these lines, I am not sure there is anyone more knowledgeable about usb and pcm than Mike Moffat (Schiit) and if 24bit 96K dac is capable, I am not sure I need to put my money anywhere else (provided i ultimately decide on USB).

 

I actually will always have both USB and enet solutions because USB is just more convenient for playback functions, and is less finicky. 

 

I also believe that "imperfect electronics" is more prominent in usb than ENET, which is why I will always have an enet solution as well for more critical listening...as has always been my contention.

 

I will always keep in mind though that if "imperfect electronics" are kept in check that all competent dacs "should" sound the same if they are doing their job...you should never have to "pay" for a dac that sounds better, the correct way it should be said, is that you pay for a dac that better compensates for "imperfect electronics" to more accurately recreate the original recorded signal....