Originally Posted by Krobar
Interesting discussion and source code on DSD to PCM conversion:
Well, if we wanted to fit a PC motherboard running Windows inside the player, this could work....
Seriously, you have to remember two things:
a) "DSD" is NOT a technical term. It is a marketing
term. The only thing that it means is whatever Sony wants it to mean on a particular day.
The only reason it exists is because the CD patents were expiring, and with them a $1 billion per year
royalty stream. They were desperate to come up with something that they could patent and license. Hence we have "DSD" or "Direct Stream Digital".
Technically, this is nothing new at all. Thirty years ago, audio A/D converters were all Successive-Approximation Register (SAR) types that required a sample-and-hold circuit. These are crude at best and can only work well with non-existent theoretically perfect FET switches and capacitors.
Then putting more and more computational power onto silicon chips got cheaper and cheaper, so everyone moved to sigma-delta converters. These oversample at a very high rate and then require downsampling (digital low-pass filters) to turn the signal into the familiar PCM (Pulse Code Modulation). So-called "DSD" is nothing more than the raw output of a sigma-delta A/D converter without running it through a low-pass filter.
But the marketing hype they created painted them into a corner. They claimed that the reason "DSD" sounded good was because it was a one-bit system, and therefore had ZERO linearity errors. Now it is true that one-bit systems have perfect linearity, but this is NOT the reason that "DSD" sounds good. The reason that it sounds good is because it doesn't require a low-pass filter on the record side, and only requires a comparatively
gentle low-pass filter on the playback side.
The problem is that you cannot
to a "DSD" signal without turning into PCM! No level changes, no mixing, no EQ, no compression, no limiting, no nothing
. The end result is that the only
way to make a disc that doesn't use PCM at some point is to do all of the mixing and EQ in the analog domain. That means that you either:
a) Record live in the studio, straight to a "DSD" recorder, with all mixing, EQ, compression, and limiting done in the analog mixing console. This is clearly impractical and has been done literally only a handful of times.
b) Record to analog tape in the normal old-fashioned way, mix, EQ, compress, and limit with analog tools down to another analog tape copy, and then transfer this analog tape to a "DSD" format. This is basically a way to re-release old analog recordings, but no new recordings are ever made in this way.
So the whole "DSD" thing is basically a fraud. Every SACD turns the music into the "dreaded" PCM format at some point during the production process. It would have been much easier and better if they had just used high sample-rate PCM to start with.
The link you posted was to a software program that would low-pass filter the "DSD" signal in two stages down to 88.2 kHz PCM. As I said in a previous post, that is not the problem. There are chips made to do this such as this:http://www.npcamerica.com/pdf/SM5819.pdf
or we could make our own (improved) filters using FPGA's. I'm not really sure what the point would be, because now you have discarded "DSD"'s mythical superiority by turning it into plain old PCM... But the problem is how do you get the signal from the SACD disc into your computer? There is no easy way as no computer ROM drive can read an SACD disc.
With the DX-5, you can connect your computer to the USB audio input and play 99.9% of all your music that way. And when you want to listen to the 17 SACD's you own, you can just stick them in the tray and play them.