From: igb@fulcrum.co.uk (Ian G Batten)
Subject: CD Technology [Was Re: Record shops (LPs)]
Organization: Fulcrum Communications
Distribution: uk
Date: Wed, 16 Dec 1992 16:31:24 GMT

In article <1992Dec15.150558.2841@lsl.co.uk> snail@lsl.co.uk writes:
> Phrase 1) Multi bit sampling CD player

The only technique up until a few years ago.  The classic chipset is the
Phillips 16x4 stuff, which is still excellent.

The samples on the disk are 16 bit resolution, with a 44.1KHz sampling
rate.  A multi-bit DAC is a switched resistor ladder followed by a
current-to-voltage converter.  Of course the problem here is that the
steps from the sample-and-hold will introduce all sorts of noise.
Output filtering is required to suppress the frequencies beyond the
sampling frequency.  Brick-wall filters in the analogue domain are very
nasty, and you have 1 octave between the top of what you want to keep at
20KHz and the bottom of what you want to remove at 44KHz.  The pass-band
ripple will be spectacular.

..therefore ``oversampling'' is used.  Zeros are introduced between the
samples and the resulting stream of original data with three zero
samples (for 4x oversampling) or seven (for 8x) or whatever are fed into
a digital filter.  The effect is ALMOST like doing linear interpolation
between the samples.  The steps are therefore narrower and, if the
resolution of the filter and DAC are better than 16 bits, probably
smaller.  Therefore the analogue filtering has to suppress signals above
4x or 8x 44.1KHz rather than above 44.1KHz.  That gives it 4 or 8
octaves between the top of the data you're interested in and the bottom
of the stuff you need to remove, rather than the 1 octave you'd
otherwise have.  But...

> Phrase 2) 1-bit sampling CD player

..building monotonic 20-bit resistor ladders is not simple.  Hence
BitStream.  The above oversampling and filtering process is used, but
far faster.  384x or 768x oversampling is used in the Phillips chipset,
for example.  Then, essentially, each sample is thresholded and a pulse
emitted if the value is greater than 50% of the peak level.  (You could
just emit the MSB, for example).  The resulting error is added back into
the following sample and the process continues.  This stream of pulses
is integrated by using it to charge a capacitor.  (That capacitor is
either on the main DAC chip or somewhere else in the so-called DAC7
chipset).

There are additional wrinkles, such as inverting the samples in the
digital domain, driving two complete DAC chipsets and then subtracting
the results.  This ``Dual Differential'' technique improves low-level
linearity --- already a strong point of BitStream.

The above describes the Phillips BitStream technique.  There are similar
techniques that use perhaps four widths of pulse, in the Sony PWM stuff.
Co-Linear DACs, such as the Burr Brown 67 chipset, do the 10 most
significant bits with a Multi-Bit and the lower bits with BitStream...

The problem with these ``lowbit'' methods is that you're suddenly
talking about MHz and above signals on the same PCB as your audio.  This
is Not Trivial.

Most consumer CD equipment is limited by the quality of the analogue
stages.  From that perspective Lowbit is A Good Thing because the output
filtering requirements are relaxed and they're a common source of
trouble.  It's also cheaper.

At the high-end, opinions vary.  Cheap BitStream seems better than cheap
MultiBit.  Beyond that it's religion.

ian