I have a CD from a Metal band called Accept - Staying a Life, it shows on the Japanese liner that it is mastered in DSD. This CD is the best sounding CD I have ever heard, You can crank the music and it seems like it just doesn't fatigue my ears, distortion seems really hard to introduce...Also, the left and right channels are really defined and extremely clear...I can play this CD on a basic sound system in my car and it still sounds really good.

Thanks, this is very true. You can hear the difference. The only digital conversion i can think of that improves the output quality.

I have in my lifetime collection about 500 cd's, some of which i thought sounded pretty good untill about 4 years ago when I was interduced to DSD. I then started buying SACD's instead, ripping them and sending to my Aurender N10. The difference to me was very noticeable to me right from the start. Recently when I was unfortunately diagnosed with Covid 19, I decided I was going to give myself a project and convert all of my CD's to DSD. I purchased some software online and worked for 6 days on the conversion process. Before long I was hearing those CD's for the first time again! They sound so much better in DSD! Don't get me wrong, my Holo May KTE makes CD's sound very good but the converted DSD files sound more analog. I will also say I have some CD's that were just bad recordings, DSD doesn't help that. They are still bad recordings! Overall, I am very happy I converted my entire collection and it was definately worth the time and effort it took to complete. The only drawback on this is storage. The DSD files take up 3 to 4 times the space on your disc drives. I have a couple of Drobo's so to me it wasn't a big deal but if you have a lot of conversions, you'll need a space to store them.

So well explained!! Thank you!!

Here's a very brief engineering explanation: In PCM, you are encoding (the "C" for code) an analog wave into a math function (generally speaking, a "Delta-Sigma" Modulation, the "M".) That code is then flashed at a clock cycle across a medium, much like Morse code (hence the Pulse.) This is received and decoded on the other end, where the wave is regenerated by inverting the math function from the previous end. Alternately, in PDM, instead of using math to encode the wave into an equation, the modulation operation is a bit different. For PDM to work, you need a clock cycle like PCM, but instead of an agreed-upon math function, we get to leverage an implicit property of nature found in calculus called an Integral. This is the idea that a series of discrete points can be used to define a smooth curve. Now, instead of pulsing a Morse code, you're playing Flappy Bird. Each flash will represent an impluse, and this will boost the signal level. This blip will then decay. If you pulse it again, it'll raise it up a little more, and it'll decay a little more, just like tapping the screen in Flappy Bird makes it bolt up, then drop down, then bolt up and then drop down. If your clock speed is fast enough, and your impulse and decay rate are calibrated correctly, it will trace out an extremely accurate wave form, though it will be "stippled," much like an inkjet printer. If you apply a smoothing capacitor, you will have a pure analog wave. Interestingly, this is very similar to the technology used to make Class D power supplies. (PWM or Pulse WIDTH Modulation uses how LONG to pulse for, not how MANY pulses.)

I would really like to hear my mic pre's through DSD. As an engineer who grew up with Neve, SSL and even DDA pre's, I have some issues with the PCM conversion in my current setup even with the Midas equipment. Would be cool if PS Audio could make a recording interface that supported DSD.

Nice pictures from Portugal in the background ;)

An interesting article, thanks Paul. I'm going through my record collection and digitizing with a view to converting to DSD128. Would it be best to capture at 176.4kHz because that is an integer multiple of DSD128 sampling frequency? Would there be any advantage/disadvantage in making the original capture at 192kHz, which is the highest rate my ADC will do?

First example

Great great question. But you didn’t actually answer it! We understand DSD is more true to life, but why converting PCM to DSD which your products do, is not explained in your answer. I use Roon to do this and it clearly sounds better and the sound stage is wider and the sound is more real.

You answered ok for me!

So, If I can't have a Directsream Dac due to my economic limitations, but have a DSD decoding dac, can I use a software to convert all my music into my HDDs or rip my cds to DSD files and then send that via USB to my dac and get better sound due to better and simpler decoding?

I had a cheap Topping DAC for a while that was buggy in converting PCM to DSD before reaching the DAC. But now I have a new DAC that does it beautifully and is a much better sounding DAC (Matrix mini i pro).

what is the best wv to DSD converter? my music sound dull/flat when converted from .WV (wave) to DSD & the bass completely disappears

DCS Audio discovered this back in the early 90s and knew it was the future of digital audio. No manufacture at the time made a dac that could handle it so they made their own convert chips specifically to work with DSD which they still use today

Alright, this got me curious so i proceeded to convert a well recorded 44.1kHz/16-bit wav/pcm file to DSF/DSD 2.8Mhz using the Tascam hi-res editor software. I am very familiar with this song and I played a specific sequence of each file over and over and over using a DSD capable DAC and for the life of me i can't tell them apart. At this point i need to be proven wrong if i state that DSD is no better at decoding to analog a PCM file that was converted to DSD than PCM is at decoding to analog the original PCM file.

new to all this Hires DSD discution...

I convert all PCM to DSD 64 with my LUMIN U1 Mini and the difference in sound coming out of the DAC is incredible!

Now am curious how that process works, converting pcm to dsd,,, what i do know is that an fm & psg soundchip produces pure analogue audio and it only uses digital co’ordinated signals to tell what and how to produce sound,, and it is possible to even simulate pcm audio on a fm & psg chip (depending on those rule restrictions wich differs per chip)by mixing all the soundchannels together and then quickly alternate the volume to create wave forms, thus achieving a low bit pcm audio in pure analogue form,because those pcm samples were beforehand converted to co,ordinated signals to control those analogue chips,,,

Why does my DAC produce better output when I upsample? Why does another DAC I have sound better when I stream DSDx2? The answer is, essentially, my software (I use JRiver) is good at doing this conversion, and the DAC is better at reproducing this input than the native 44.1khz/16bit. There may be any number of reasons for this, but the point is that DACs are not perfect, and that is why you do such conversions.