Haro? More like HERO!

Yes

@PimpinBassie2, DSD is that magic signal that can be picked up as analogue or digital. It does away the need to know byte size, bit alignment, signed/unsigned, CRC (checksums/parity/redundancy bits).

True. And engineering is awesome for knowing such things. Still the curious mind wants to observe, we want to observe and see if there is a perceivable difference in these waveforms.... I giggle at it. I'm an engineer and have audiophile friends and enjoy the topics. But the masses decided for us when we started downloading and streaming lossy codecs and listening on lo fi systems like earbuds and bt speakers.... Convenience won over fidelity and that was obvious..... So these topics are left to us nerds as a purely intellectual pursuit.

it's not really propitiatory anymore tho

Its both Sony and Phillips.

It only takes 1 bottleneck to ruin the whole chain of audio preservation. Bad mic on the studio, bad raw files storage, bad mastering that overly process the signals.. restricted audio formats, then restricted audio equipment, to finish off with bad ears capabilities.

We already do. Ogg, flac, and the open document format.

this, plus funnilly enough (DSD) you could de-ess a bit also. Good vibes tho thanks

Most record in 24bit so 32bit recordings will sound slightly better, especially if you convert to the higher DSD formats

Do the math. Every bit of depth cuts the signal level in half. If you think of a 2V line level sgnal, 2 bits would give a 1V resolution between steps. 3 bits .5V. 24 bit would require resolving roughly 0.00012V per step. Technically impossible today. 20 bit is about today's limit. Sampling rate is far more important than bit depth.

Like loudness compressed PCM, DSD increases the dynamic range and softens it out with it's higher sample rate, even with conversions.

I don't know about that, but I just setup DSD, and I will confirm, there is a difference, and it's not subtle. If the technical limitations are overcome, DSD will replace PCM. It sounds that much better. To give an analogy, when DSD first kicked in, it was like water cleared from my ears. I didn't even know PCM sounded so veiled until I heard DSD.

then the correct sound setting in the computer is also important. yes apparently you should have a program called bite perfect as well

Sure the original DSD rate was 2.8MHz, but there are multiple implementations, which most DSD DACs have. DSD512 is 24MHz. As far as I understand it, that's the general idea of how DSD works, which parts of it are very inaccurate?

Lior, it's not so much as inaccurate as simplified. DSD is hyper precise and is, at it's core, dumping the raw signal from the AD/DAC into a wrapper. Hardware recorders can set the DAC to the right frequency to the DSD depth you select that's why it's not really a consumer format. You need hardware that operates differently then the set rate PCM DACs, much of what PCM recording does is in software, hense cheaper then the variable frequency DACs used for DSD.

@@mamaluigi0631 Yup

the DSD explanation is hilarious. Too much fun for one day.

High praise, thank you!

DXD is PCM, but 24-bit 352.8KHz studio format. But dsd is marketing. There are almost no pure DSD recordings available to consumers. This is because there is currently no way to edit, mix, and master DSD files. Only the rare DSD recordings made from a mixed and mastered analog recording, or those recorded direct to DSD without any postproduction, are pure DSD. Most so-called pure DSD recordings are, in fact, edited in PCM. Most commercially available DSD recordings have to be converted back and forth to a PCM format in order to do postproduction editing, mixing, and mastering. In each of these conversions, more quantization noise and/or quantization errors are added to the recording. This leads many to ask: why degrade performance by adding the additional step to convert to DSD when the master is already in PCM?. DSD has significantly higher quantization noise than PCM, and the noise is much closer to audible frequencies, requiring significantly more sophisticated digital filters, as well as noise-shaping and upsampling algorithms, that can result in distortion of the analog signal. When a PCM file is played on a DSD or Bit Stream converter, the DAC chip has to convert the PCM to DSD in real time. This is one of the major reasons people claim DSD sounds better than PCM, when in fact, it is just that the chip in most modern single-bit DACs do a poor job of decoding PCM. Another issue is 96KHz and 192KHz HD audio. Though modern sampling rates are high enough to fool the human ear, quantization errors still occur when translating from one format to another. For example, when Sony decided to archive their analog master libraries to DSD64 back in the mid 1990s, they were wrong to believe that these masters would be future-proof and able to reproduce any consumer format. The fact is, these masters could only properly reproduce a format that was divisible by 44.1KHz. So any modern 96KHz or 192KHz recording created from DSD64 master files have quantization errors. This leads me to one of the many things that enrage me about the recorded entertainment industry. If 44.1KHz was the standard that was engineered to put aliasing errors in less critical audio frequencies, then why did they start using multiples of 48KHz?!?!?!? All they had to do was go with 88.2KHz and 176.4KHz as the modern HD consumer formats, and all of this mess could have been avoided. The same thing happens with DXD (24-bit 352.8KHz) equally divisible by 44.1KHz. What blithering idiot decided to put a wrench in the works with 96KHz and 192KHz HD audio?

Nice wall of text, haha. Very interesting stuff here, thanks!

You're absolutely spot on Javier. Very well said!

I see DSD as a Mastering and Archiving format, even PCM 192khz 24bit converted to DSD gives the PCM enhanced depth, being able to distinctly identify specific instruments that was muddied when in PCM. DSD is also coming back with the Audiophile scene and you can buy new 2017/2018 built recorders and even the old professional recorders are keeping value. What became obsolete is SACD, what survived is DSD and expanded upon with DXD to get even more from the format. What people don't understand is DSD is basically a wrapper of the RAW signal before the A/DAC samples it. So your getting basically a direct digital conversion from analogue which retains nearly all of analogue's unique characteristics, like the high frequency noise, which is above 20khz and out of the hearing range of 99% of the population btw. Jay did good on the basics, but his perspective in more of the every day user, if he explores more into DSD he'll have a better overall perspective.

Ok, then explain it to us if you know it

Steve Sene I think he was being sarcastic

Çerastes is correct. Whether sampling 1 bit at a time or 24, the end result is the same: an output that's practically the same as the input. 1+1+1+1... or 1+2+4+8 is of no consequence since the sum will be the same in both cases. If the sum is different, it's a design error, not some "quality" of one or the other methods of addition.

The difference is precision and sound stage, DSD is far from obsolete, there's almost zero quality loss from the actual source, whether from a microphone or vinyl. There's a perceptional difference between instruments and dynamic range. If you listen to reference level headphones or speakers it's very noticeable. Less so on KZfaq because the transcoders degrades it heavily. Get Foobar2000 with the DSD plugin and download an actual dsf or dsd file and i'll guarantee you can hear the difference. Unless you actually hear it and compare it you don't understand it nor qualified to brush it off.

Let's be fair now. Precision is a thing, and a thing that I am familiar with. It's just that saying "precision" unconditionally has no meaning in itself. James, what parameters are you claiming DSD is more precise with? What is the difference in precision, and why is it significant? Please be specific. Precisely what "quality" are you claiming that DSD doesn't lose? How does LPCM lose the same quality, and why? Which "actual source" are you referring to? You never defined any of those! I do hope you aren't claiming that DSD bits are hand-made and "artisanal". BTW, what is your reference level? As for your entreaty to install software and listen to things download, no thanks. You may not know this, but computers inherently use the same type of data structures as LPCM does. If that's how you've been listening to "DSD" then you've been fooled; all this time you've been experiencing the superior precision, sound stage and quality of LPCM. If you have an all-serial signal chain to lend, I'd be happy to make an apples-to-apples comparison.

Çerastes, I recommend You following experience - very good turntable with good tonearm, very good moving coil low output cartridge, step up transformer and good phono stage. Now let's make vinyl rip with digital recorder KORG MR-1000 to 1 bit 5,6MHz dsf file (PDM DSD). This file can be 3-5 GB big. Later make PCM 16bit 44,1kHz rip flac or wav file with the same or similar digital recorder like Korg, Tascam. File will be much smaller 300-500MB max. Later compare both files on very good hi fi/hi end systems. Difference is HUGE. DSD is the best technology to digitalize analog sound sources. For instance sound of violin is different, sound of man's voices, percussion, cymbals etc. Try to do it. Best regards from Poland.

Lukasz P., if there is a "difference" then the whole A-D/D-A process has failed miserably. What you should be getting is the same, not different. Funny how, whenever I ask people to back up their claims with substance, I get radio silence...