iXBT Labs - Computer Hardware in Detail






Digit Life

This is the first editorial article for this site. Such editorial reviews will issue periodically (I hope at least once a week). The aim is to carry pleasant for reading and easy for comprehending information on a free subject (that is, to my taste). Undoubtedly, the topic has to relate to the site's subject-matter. At least, to some degree.

There I will consider a technology so far untouched in the computer fields which still can influence habitual conceptions strongly. And that is digital representation of everything out there, and different methods of its processing.

The SACD standard appeared due to a simple commercial reason. The date of payments under a licensing agreement to CD-Audio technology developers was approaching, and the income received by them was incredible. According to rumors, it was comparable to that received by the developers from sale of CD equipment of their own production. A way-out is obvious: Sony and Phillips were to create a new format capable to force out CDs. And mainly to attract comparable license assignments. That was a prehistory to SACD creation (Super Audio CD). Obvious it's impossible to attract customers and companies by a simple increase in capacity - the whole commercial sound recording sticks to 74 minutes and capacity increase isn't worth realization. And there lies only one way - to gather additional money for a 2-CD set of classical music. But who is to blame that at that time there wasn't 74-minute limitation. Size is out of discussion, so it's necessary to develop completely new quality. And it shouldn't be just an intensive jump up (we have got already DVD with 192 KHz 24 bit), but a qualitative new approach.

Many modern, decent AD and DA converters are build on a so called 1 bit scheme. If I had had my way I'd have called it the "digital PDM scheme", but I wasn't afforded such opportunity =). An analog signal is represented in the digital form with the help of PDM (Pulse Density Modulation) with a variable quantization step. Pulse duration defines the level of an analog signal. To be exact, it is defined by a ratio of the pulse duration to the duration of the following pit (see the graph).

That is the energy transmitted in one period of a decoded PDM signal. It gets clear that it's not difficult to make a DA converter for such signal. Just take a capacitor and let it integrate for some definite time period. No doubt that a real model is more complicated, but it's based on the same principle. Besides, the ADC has a perfect ratio price/quality. Conversion is fantastic and the most problems concerning parallel and other DAC/ADCs disappear. For example, this scheme is adaptive: if a signal has a small amplitude the readings go more often and detailing increases. This, in its turn, completely corresponds to well known nuances of sound perception by a human being.

In modern Hi-Fi facilities such convertors are used widely. But, unfortunately, information is kept in linear code with a fixed number of bits per reading. It means that after such ADC it's necessary to convert 1-bit PDM into a standard one, say 16 bit, with a fixed period of reading selection. To save possibly compressed (with losses or without), to read, decode (unpack), reconvert into successive PDM, transfer to DAC. Difficult. And the main thing is that intermediate conversions bring in their own distortions. Good (with low distortion level) intermediate converters cost expensive, nullifying all advantages of 1-bit converters. The way-out should conclude in getting rid of intermediate conversion. To transfer, compress, store and process a successive PDM stream i.e. in the form of a bit stream.

Of course, SACD developers followed this way. Recording density increased several times as compared with CD. Judging from today's point of view, it's ordinary. The bandwidth went up as well from 44'100*16=705'600 bit/s to 705'600*4=2'822'400 (for each channel). PDM stream was divided into bits with sampling frequency around 2.8 MHz (the same 2822400 bit/s). It's called DSD (Direct Stream Digital). And then recorded on a CD. That's all. The others are just details. For example, a bit wider aperture for an optical system provides better scratch prevention (though just a little). And the most important that it provides an opportunity to create two-layer discs which are seen by a SACD drive as SACD, and by A CD drive as usual CDs.

Further, if there is some information, one feels like compressing it and converting into prevalent formats. The latter can be executed simply (quantization step of the PDM stream is divisible by standard 44'100), and the former requires new compression methods (with losses and without) oriented to digital bit streams. Believe me, they have been "living" for a long time already and quite successfully. For example, on the base of associative or "entropic" coding. Some of them are realized on hardware level without any problems.

Why don't we see similar technologies on PC? Who prevents installing 1-bit convertors on sound cards? There we can save on board layout - data on codecs are transferred successfully... But no - there tells upon a parallel orientation of modern architectures. It's inconvenient (for architectures, and therefore programmers) to work with successive bit streams. Although some are doing it. There arise nondestructive archivers and channels of successive data transfer. I can only hope that in the near future music will be kept and processed (right up to usage of different filters and effects) in successive DSD-like formats.


Bandwidth 0-100 KHz 0-20 KHz
Dynamic range 120 dB 96 dB
Sounding time 110 m 74 m


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