Hey, cool topic J, sounds pretty nice to me. I wish we lived near each other.
I have the MR2000S DSD coming this fall. I plan on using it as a replacement for my old... very old DAT and recording a lot of live stuff. I'm more excited after reading this topic and hearing your track.

Why did you buy yours and what do you love the most about these?

Thanks for sharing this. (y)

This is a very interesting topic indeed. So much to learn!

Codemonkey was kind enough to give a basic Idea, but I really need to dig deeper into this concept.

It would be a very interesting PCB layout that had both analog signals and 5.6Mhz clocks bouncing though it.

At least the physical size of the filters would be smaller.

Does anyone have any technical links on the process? I did some searching but found only marketing crap.

This indeed is an interesting topic, and one for which there is much mis-information floating around.

A quick overview of the principle of operation of single-bit audio is as follows: imagine a square wave switching between 0V and 3V. Considering average values of the wave, because it is symmetrical, the average value is constant at 1.5V and contains no information. Now if you modify the wave to leave the 0-to-3V transitions in the same place but shift the 3-to-0V transitions backwards and forwards in time about their original position, the average value varies. Make the time-shifting vary with your input signal, then the average value of the output will be a representation of the input signal. In most versions of single-bit systems, both transition directions (high-to-low and low-to-high) are shifted.

The "1-bit" or "single-bit" name comes from there being only two levels (0V and 3V in our example), which in amplitude terms can be represented by the state of a single bit. All the signal information is contained in the times at which that bit changes state and not in its amplitude..

From this can be seen that the limit to the resolution of a single-bit audio system is the coarseness of the time instants at which decisions are made whether or not to change the state of the single bit. In principle, the higher the sampling rate, the higher the resolution. In practice, there are numerous other effects to consider to make a usable system, but the basic principle of operation is as described.

Incidentally, the front end of a single-bit system is identical to an over-sampled PCM system. The single-bit (DSD) stream is extracted at an earlier stage in the digital filtering process than for PCM. Also, the mix of precision analog circuitry and high-speed clocks is not new for DSD, and has been a major challenge in high-precision converters for many years. My day job includes designing these types of systems and making them work in audio and industrial fields.

It appears I was spectacularly wrong. Sorry link.

No worries Codemonkey. I still think your idea is a cool one.

Thanks Boswell so it sounds a bit like pulse width modulation triggered by the input audio signal? This is probably not true but it does sound bit like it.

Incidentally, the front end of a single-bit system is identical to an over-sampled PCM system. The single-bit (DSD) stream is extracted at an earlier stage in the digital filtering process than for PCM.

So the front end filtering could be reduced for a purely DSD system?

In practice, there are numerous other effects to consider to make a usable system, but the basic principle of operation is as described.

What types of other issues have your designs encountered?

Thanks again boswell, just trying to understand.

Link555 wrote: So the front end filtering could be reduced for a purely DSD system?

Not usefully. The filtering I talked about is digital filtering, primarily a cascaded integrator comb filter, often found inside the ADC chip but can be done by external DSP. [[url=http://[/URL]="http://en.wikipedia…"](Reference)[/]="http://en.wikipedia…"](Reference)[/]

Link555 wrote: What types of other issues have your designs encountered?

Any design has to work to a specification and within a list of constraints that includes a manufacturing budget. Some of the most difficult design constraints are not directly audio related, such as electromagnetic compatibility (EMC), power supples, mechanical housing, and testability, not to mention ease of manufacture and having to second-guess the availability of all the chosen components for at least 5 years after a design goes into production.

At the audio level, the big challenge is trying to achieve the noise and distortion figures demanded by the design specification, and this is where mixed-signal problems (low-level audio co-existing with high-speed digital circuitry) can occur. Even simple things like the exact point at which analog and digital grounds are inter-connected can have a big impact on the resulting performance.

Open up a cheap piece of audio gear and also a well-respected high-quality piece of a similar function, and at first sight they may appear to be much the same: a printed circuit board with I/O jacks and some user interface controls. Examine more closely, and you may see that the high-quality equipment uses 6- or 8-layer circuit boards, the boards are several revisions into their history, the power supply is well separated, screened and has separate looms for digital and analog power. It will smack of attention to detail. This is the only way to get quality results.

Thanks for the response boswell do you know of any good references for the way 1-bit ADC's work?

Link555 wrote: Thanks for the response boswell do you know of any good references for the way 1-bit ADC's work?

[="http://www.korg.com/ClassDetail.aspx?ID=94"]This[/]="http://www.korg.com…"]This[/] is an article by Korg outlining their 1-bit technology.
[[url=http://="http://dafx04.na.in…"]Here[/]="http://dafx04.na.in…"]Here[/] is a more interesting paper from a group at QMC in London proposing various methods of applying EQ and effects to single-bitstreams. [Page 2 does not show up on my PC; you may have better luck]

Thanks again Boswell. The filtering aside Is there a site you know of, that talk about the actual ADC conversion aside from the filtering section?

Link555 wrote: Thanks again Boswell. The filtering aside Is there a site you know of, that talk about the actual ADC conversion aside from the filtering section?

Try [="http://www.maxim-ic.com/appnotes.cfm/an_pk/1870"]this[/]="http://www.maxim-ic…"]this[/] from Maxim. Think of DSD data being taken out of the ADC at the point where the article shows "1-bit data stream". The details of a true DSD converter are slightly different, but the principles are similar enough for this to be a useful description.

The DSD Pro Audio site has an [[url=http://="http://www.dsdproau…"]explanation page [/]="http://www.dsdproau…"]explanation page [/]

For what it's worth, there is also a [[url=http://[/URL]="http://en.wikipedia…"]Wikipedia page[/]="http://en.wikipedia…"]Wikipedia page[/] with a jumble of information about DSD.

Thanks! The maxim link is exactly what I was looking for!

In your work have you designed Sigma-Delta ADCs at chip level?

Link555 wrote: Thanks! The maxim link is exactly what I was looking for!
In your work have you designed Sigma-Delta ADCs at chip level?

No, unfortunately. Mixed-signal chip design is a very specialized skill that I regret I have never had the opportunity to acquire. However, I have used many of the different ADC architectures in my equipment designs including the type where you can get in and re-program the internal digital processing to adapt the basic design for use in new and sometimes weird applications.