Why does it say 1-bit Dual D/A converter on my CD player?

To understand the reason for a 1-bit Dual D/A converter, it
is helpful to know a little about the digital-to-analog
conversion process. In a CD (and any other digital recording
technology), the goal is to create a recording with very high
fidelity (very high similarity between the original signal
and the reproduced signal) and perfect reproduction (the
recording sounds the same every time you play it no matter
how many times you play it). To accomplish these two goals,
digital recording converts the analog wave into a stream of
numbers and records the numbers instead of the wave. The
conversion is done by a device called an analog-to-digital
converter (ADC). Then, to play back the music, the stream of
numbers is converted back to an analog wave by
a digital-to-analog converter (DAC). The analog wave produced
by the DAC is amplified and fed to the speakers to produce
the sound.


When you sample the wave with an analog-to-digital converter
you have control over 2 variables:

* Sampling rate - controls how many samples are taken per
second
* Sampling precision - controls how many different
gradations (quantization levels) are possible when taking the
sample

In the following figure, let's assume that the sampling rate
is 1,000 per second and the sampling precision is 10:

The green rectangles represent samples. Every 1/1000th of
a second, the ADC looks at the wave and picks the closest
number between zero and 9. The number chosen is shown along
the bottom of the figure above. These numbers are a digital
representation of the original wave. When the DAC recreates
the wave from these numbers, you get the blue line shown in
the following figure:

You can see that the blue line lost quite a bit of the detail
originally found in the red line, which means the fidelity of
the reproduced wave is not very good. This is the sampling
error. You reduce sampling error by increasing both the
sampling rate and the precision. In the following figure,
both the rate and the precision have been improved by
a factor of 2 (20 gradations at a rate of 2,000 samples per
second):

In the following figure, the rate and the precision have been
doubled again (40 gradations at 4,000 samples per second):

You can see that as the rate and precision improve, the
fidelity (similarity between the original wave and the DAC's
output) improves. In the case of CD sound, fidelity is
an important goal, so the sampling rate is 44,100 samples per
second (44.1 KHz) and the number of gradations is 65,536. At
this level, the output of the DAC so closely matches the
original wave form that the sound is essentially "perfect" to
most human ears.

The DAC typically uses a different resistor for each bit.
A 4-bit DAC needs 4 resistors working in parallel to provide
a steady analog signal. When you get to the 16-bit or even
32-bit level found in CDs and DVDs, the number of gradations
required per resistor makes it very difficult to precisely
match values. For example, a typical 16-bit DAC would have 16
resistors requiring a total of 65,536 gradations.

What a 1-bit Dual D/A converter does is allow the
digital-to-analog conversion to happen without the need for
all those extra resistors. Essentially, this type of DAC does
not use a bank of resistors operating in parallel. Instead,
it creates a carefully modulated signal from the digital. The
converter relies on noise shaping, a phenomenon that takes
advantage of the human ear's inability to notice noise when
it occurs in higher frequencies. Basically, the human ear is
most sensitive to noise at 5 KHz, and is almost unable to
detect it at 20 KHz.

A key part of the converter is a circuit called a delta-sigma
modulator, which takes the binary signal (1s and 0s) from the
CD and changes them to a steady pulse, called a pulse train.
The pulse train contains an average of the change in the
amount of energy represented in the sample. A low-pass filter
removes all time-domain information and recovers only the
average energy of the pulse train that feeds it. The key here
is to understand that the pulse-train waveform is clocked at
a very high frequency compared to the 44.1 KHz sample rate.
The pulse train is sent through the DAC and changed into
an analog signal.

The delta-sigma circuit has two main sections:

* Delta receives the incoming digital signal and monitors
the outgoing pulse train. It creates an error signal, which
is based on the difference between the binary signal coming
in and the pulse train going out.
* Sigma adds up the results of the error signal created
by delta and supplies this sum to the low-pass filter.

The error signal is used by the low-pass filter to average
the analog signal. Basically, this means that minute
adjustments are made to the analog signal to compensate for
the differences between the binary signal and the pulse train.