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Worlds Greatest DAC and what it does differently


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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.

 

 

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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".

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1 hour ago, beerandmusic said:

 

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".

 

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.

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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.

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2 minutes ago, beerandmusic said:

 

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.

 

Compliment received and acknowledged 😜

 

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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.

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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)

 

 

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What's "VASTLY DIFFERENT" is the subjective sense of what you perceive, when listening ... worrying about a 1kHz waveform is pointless, because it has zero emotional content, or structure which is meaningful in a musical sense.

 

Miska is correct ...digital replay just turns out to be very fragile to a huge range of inteference factors - and the anomalies heard, quite easily, from this are what leads to, umm, "vast differences".

 

The "best" DACs, from somebody like MSB, are engineered to an extreme degree, to mitigate this interference and noise - and yield the best subjective performance.

Frank

 

http://artofaudioconjuring.blogspot.com/

 

 

Over and out.

.

 

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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.

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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.

 

 

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24 minutes ago, beerandmusic said:

i laid out clear assumptions::

1.  All hardware (besides DAC) is the same

 

Yeh he said “Point is that the analog output from both DACs is not the same”

 

He’s not talking about what differences can and can’t be heard. He’s saying he can show measurements that the DACs will have different analogue outputs.

 

There’s nothing controversial about this.

 

The controversy is all in what differences can and can’t be heard (when looking at measurements above 20kHz) but I’m not going there. That topic has been bashed on every audio forum for years.

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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?

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1 hour ago, beerandmusic said:

 

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.

 

My point is that I very much doubt you will ever measure anything significant by what happens to a 1kHz waveform - only by setting up quite a sophisticated test layout - and, no, I don't know what that would be - would one have a chance of understanding what is the nature of the underlying distortions, etc ... In the meantime, I use my ears, with music ... 😉.

Frank

 

http://artofaudioconjuring.blogspot.com/

 

 

Over and out.

.

 

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18 hours ago, beerandmusic said:

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

 

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.

Math_a.thumb.png.48a44818dd6392130e7ca779bc191b11.png

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.

16bitPCM_and_24bitPCM_a.thumb.PNG.c990d9cd2f951257fc9e2cbaf7d333e3.PNG

 

First part of the sampled values plotted using Audacity:

1kHzSine_PCM_samples_Audacity.thumb.png.9da36b0d53d52773c97743a4bf365955.png

 

DAC output analog waveform simulated by Adobe Audition:

1kHzSine_PCM_samples_and_reconstructed_waveform_Adobe_Audition.thumb.png.2bf2a5fa85394b5794fb1c9accd0d012.png

 

 

Sunday programmer since 1985

Developer of PlayPcmWin

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18 minutes ago, barrows said:

My feeling is that there is some "magic" with DSD 256 (and perhaps above).  But of course this is not really "magic" at all.  High sample rates, including high rates of PCM-like quality allow for relatively simple actual conversion paths.  This is how most Sigma Delta style DACs work.  Tthese high rates also allow for even discrete (non chip based) converters, which give designers additional freedom (if they are clever enough) to develop a unique converter topology which may offer advantages over standard chips.

For chips, when one looks at, say, ESS-it oversamples to a very high rate (similar to DSD rates) and 5, 6, or 7 bits, and then converts to analog from this very high rate.  dCS uses a similar approach, but they do the final conversion into analog using a discrete converter.  Chord, again uses a similar approach, again using a (similar to dCS) discrete final converter stage.

Jussi's (AKA miska) DSC-1 DAC design, now adopted in various different iterations by Holo Audio, Denafrips, T+A, among others, is a DSD based discrete converter stage designed to be fed by a very high rate single bit signal, generally DSD 256 and above.

Mola Mola, in the excellent Tambaqui DAC, uses a variation on the discrete converter theme as well, with the final converter stage being a multi-element switch and resistor set up, similar to DSC-1.  Same with Playback Designs, and I suppose Emm Labs...

I am listening as I type to Bricasti M3 DAC, which uses a different, discrete converter stage design of their own, essentially also based on a high speed switch and filter, for single bit DSD input only.

 

Most of these different DACs sound different, but I have experience now with 4 different single bit based DACs, all of which have used some variation on a discrete converter stage.  All of them have seemed to share, in differing degrees, a quality of conversion which sounds more natural, and easy going to me.  I do not mean that they are rolled off, or "soft" sounding, but that they sound more like music, and less like an electronic representation thereof.

 

There are a lot of really good DACs out there these days, but for me, I do seem to prefer the DAC which use discrete conversion approaches, with high rate DSD input.  This can be done on a chip as well, as with AKM DAC chips' "direct DSD" path option.  But, when a designer goes discrete, they have control of the performance, and they may be able to do better than what is available on a generic DAC chip (or not, as the case may be).  

Are there source side benefits? I'm told the Mojo doesn't really do DSD, converts to PCM but using Stylus/Roon sounds cleaner to me with DSD 128/256 upsampling than max PCM rate.

Regards,

Dave

 

Audio system

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Just now, davide256 said:

Are their source side benefits? I'm told the Mojo doesn't really do DSD, converts to PCM but using Stylus/Roon sounds cleaner to me with DSD 128/256 upsampling than max PCM rate.

I am not sure what you are asking....

Chord DACs, except DAVE, all convert incoming DSD to PCM 705.6 before then upsampling to very high (MHz rates) at, as I recall 5 bits.  the this data goes to their discrete conversion stage.  I would not send dSD rates to a Chord DAC, as it will be down sampled to PCM 705.6 anyway.  If you want to oversample in software with a Chord DAC (again excepting DAVE), I would send PCM 705.6 to the DAC, as this will bypass some of the DACs onboard processing.  Note that this what the mScaler does...

SO/ROON/HQPe: DSD 256-Sonore opticalModuleDeluxe-Signature Rendu optical--Bricasti M3 DAC--DIY Purifi Amplifier-Focus Audio FS888-JL E 112 sub-Nordost Tyr USB, DIY EventHorizon AC cables, Iconoclast XLR & speaker cables, Synergistic Orange Fuses, Spacetime system clarifiers.                                                       

                                                                                           SONORE computer audio

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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.

Math_a.thumb.png.48a44818dd6392130e7ca779bc191b11.png

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.

16bitPCM_and_24bitPCM_a.thumb.PNG.c990d9cd2f951257fc9e2cbaf7d333e3.PNG

 

First part of the sampled values plotted using Audacity:

1kHzSine_PCM_samples_Audacity.thumb.png.9da36b0d53d52773c97743a4bf365955.png

 

DAC output analog waveform simulated by Adobe Audition:

1kHzSine_PCM_samples_and_reconstructed_waveform_Adobe_Audition.thumb.png.2bf2a5fa85394b5794fb1c9accd0d012.png

 

 

 

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.

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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.

Math_a.thumb.png.48a44818dd6392130e7ca779bc191b11.png

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.

16bitPCM_and_24bitPCM_a.thumb.PNG.c990d9cd2f951257fc9e2cbaf7d333e3.PNG

 

First part of the sampled values plotted using Audacity:

1kHzSine_PCM_samples_Audacity.thumb.png.9da36b0d53d52773c97743a4bf365955.png

 

DAC output analog waveform simulated by Adobe Audition:

1kHzSine_PCM_samples_and_reconstructed_waveform_Adobe_Audition.thumb.png.2bf2a5fa85394b5794fb1c9accd0d012.png

 

 

 

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...

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14 minutes ago, barrows said:

I am not sure what you are asking....

Chord DACs, except DAVE, all convert incoming DSD to PCM 705.6 before then upsampling to very high (MHz rates) at, as I recall 5 bits.  the this data goes to their discrete conversion stage.  I would not send dSD rates to a Chord DAC, as it will be down sampled to PCM 705.6 anyway.  If you want to oversample in software with a Chord DAC (again excepting DAVE), I would send PCM 705.6 to the DAC, as this will bypass some of the DACs onboard processing.  Note that this what the mScaler does...

The question is

 

assuming  adequate processor power and memory with competent software, which will be less impacted  as USB source by issues in the source hardware and software,

1 bit xMhz DSD output or 16/24 bit 705khz PCM output?

Regards,

Dave

 

Audio system

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