The Series 2 Digital Audio Adapters are IBM AT compatible add-on
boards which convert high fidelity analog signals to digital data for
storage to, and retrieval from, disk.
The Series 2 adapters sample two channels of audio from 6.25
kHz to 50kHz with 16 bit resolution. They incorporate Sigma Delta
technology with 64 times oversampling, providing superior fidelity
at greater than 90 dB signal-to-noise ratio.
ABOUT DIGITAL AUDIO
In professional circles, digital audio has been with us for over 10
years. With the advent of the compact disk in 1983, digital audio
has become commonplace as a consumer item. Few will argue
that digital audio has afforded an order of magnitude improvement
in overall sound quality and signal-to-noise ratio over the best
analog systems which preceded them. But just what is digital
audio, and where and how is it used?
It is possible to use digital data transmission techniques to
transmit digital audio signals by wire or radio. However, this
practice has not yet become common due to the extremely wide
signal bandwidth required to transmit real-time digital audio
signals. For the present, digital audio techniques seem largely
confined to the recording and playback of music and other audio
signals where, in a few short years, digital audio technology has all
but replaced the previous analog record/playback techniques. In
the present decade we will see digital audio technology replace
analog technology in most signal processing functions in both the
professional and consumer markets. It is also likely, particularly
with the advent of fiber optic cables, that digital audio technology
will be utilized in the transmission of real-time audio signals on a
widespread basis.
But what is digital audio?
1
In essence, digital audio is a technological process whereby an
analog audio signal (produced when sound waves in the air excite
a microphone) is first converted into a continuous stream of
numbers (or digits). Once in digital form, the signal is extremely
immune to degradation caused by system noise or defects in the
storage or transmission medium (unlike previous analog systems).
The digitized audio signal is easily recorded onto a variety of
optical or magnetic media, where it can be stored indefinitely
without loss. The digitized signal is then reconverted to an analog
signal by reversing the digitizing process. In digital audio
record/playback systems, each of these two functions is
performed separately. In digital audio signal processing systems
(where no record/playback function occurs) both analog-to-digital
and digital-to-analog conversion processes occur simultaneously.
A variety of techniques are possible, but the most common
method by which audio signals are processed digitally is known as
linear pulse code modulation, or PCM. Let's take a brief look at
how PCM works.
Converting an analog signal to digital is a two-stage process,
sampling and quantization. This is illustrated in Figure 1. At regular
intervals, a sample-and-hold circuit instantaneously freezes the
audio waveform voltage and holds it steady while the quantizing
circuit selects the binary code which most closely represents the
sampled voltage. Most digital audio is based on a 16-bit PCM
system. This means that the quantizer has 65,536 (216) possible
signal values to choose from, each represented by a unique
sequence of the ones and zeroes which make up the individual
code "bits" of the digital signal.
The number of these bits generated each second is a function of
sampling rate. At a relatively low sampling rate of 8 kHz (suitable
for voice) far fewer code bits are produced each second than, for
example, at the 44.1 kHz sampling rate used for commercial
compact disks. For a two-channel stereo signal at a 44.1 kHz
sampling rate, some 1.4 million bits are generated each second.
That's about five billion bits per hour,which is why you'll need at
least an 800 Megabyte hard disk to record an hour of compact disk
quality music.
2
Figure 1: Analog-to-Digital Conversion
To visualize the analog-to-digital conversion process, refer to
Figure 1. At the top is one cycle of an analog input signal wave.
We've used a simple sine wave to make visualization easier. In
this example, the signal has a peak-to-peak amplitude of 20 units,
measured by the scale on the left. The sampling frequency is
many times higher than the signal being sampled and is shown
along the bottom of Figure 1. Once for each cycle of the sampling
frequency, the sample-and-hold circuit "slices" the input signal,
allowing the quantizing circuit to generate a (digital) number equal
to the closest (of the 65,536 possible discrete values) quantization
value of the input signal at the time the sample is taken. This
repeats for each successive cycle of the sampling frequency and
the quantizer generates a continuous "bit stream" which
represents the quantized signal. The continuous stream of digital
audio information is converted into a digitally modulated signal
using a tech nique known as linear pulse code modulation.
Digital-to-analog conversion (used in playback) is the exact
opposite of the analog-to digital conversion process and is
illustrated in Figure 2.
3
In digital-to-analog conversion, the PCM bitstream is converted at
the sampling frequency to a continuously changing series of
quantization levels which are individual "steps" of discrete voltage
equal to the quantization levels in the analog-to-digital process.
The shape of this continuously changing stream of quantization
levels approximates the shape of the original wave. This is shown
in the top half of Figure 2. This signal is then passed through a lowpass filter, which removes the digital "switching noise." The end
result, shown in the bottom half of Figure 2 is an analog output
signal whose waveshape is a very close approximation of the
original analog input signal. The foregoing is a very brief and, of
necessity, oversimplified explanation of how digital audio works.
For the interested reader, the book Principles of Digital Audio by
Ken C. Pohlmann, copyright 1985 by Howard W. Sams, is highly
recommended.
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Figure 2: Digital-to-Analog Conversion
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MINIMUM HARDWARE RECOMMENDED
?16 MHz 386-SX or compatible
?28 mSec average access hard disk
?1:1 Interleave hard disk controller
?Mouse
?VGA display
ADAPTER INSTALLATION
Make sure the main power to your computer is OFF. You will need
a full-size, 16 bit/AT slot. If you are unfamiliar with the internal
design of your computer see its "Guide to Operations" manual for
step by step installation procedures.
Read JUMPER SETTINGS and for information about configuring
the adapter before plugging it into the slot.
JUMPER SETTINGS
These SX series adapters have four hardware jumpers. They are
JP1, JP2, JP3 and JP4, which are used for multiple board
operation. JP1 should be installed for a single board system.
NOTE: On the SX-7, jumper JP2 serves as the adapter selec tion
jumper. Adapter 1 is at the top, with adapter 4 at the bottom.
1
2
PLACE JUMPER HERE IF YOU HAVE ONLY ONE
ADAPTER
3
4
6
I/O ADDRESSES AND INTERRUPTS
The valid I/O addresses for the SX7, SX9, SX11, SX-12a, SX20,
SX22, SX23e and SX26 are:
180h, 22h, 280h, 300h, 320h and 380h
The valid interrupts are:
2, 3, 4, 5, 10, 11 and 12
SX-12a, SX-20
There are 5 external connections located on the front surface of
the audio board's metal bracket (Fig. 3).
Audio Input:
Line: RCA jacks, 2VRMS/+6dBV max (digital clipping),
with an impedance pf 20 k ohms
Audio Output:
Headphone: 1/4" stereo phone jack, 0.5 VRMS into 8
ohms
Line: RCA jacks, 2VRMS/+6dBV max (digital clipping),
with an impedance of 470 ohms and a load
impedance of >10k ohms
SX-7
There are four external connectors located on the SX-7 audio
board's metal mounting bracket. See Figure 4. There is also one
three pin header (JP1) for an optional user-supplied headphone
connection, which would allow for a front panel headphone jack.
7
SX-12a/SX-20
Balanced Analog I /O Connector
678
9
RIGHT LINE IN
LEFT LINE IN
HEADPHONE OUTPUT *
RIGHT LINE OUTPUT
LEFT LINE OUTPUT
* Stereo Headphones
Figure 3: SX-12a, SX-20 Connector Locations
1
LEFT
RIGHT
GND
JP1
SX-7
DB-9 Balanced Out
Headphones
Unbalanced Right
Unbalanced Left
Pin Assignment
1 Gnd
2 nc
3 nc
12345
DB-9
Female
4 Right Out 5 Left Out 6 nc
7 nc
8 Right Out +
9 Left Out +
Figure 4: SX-7 Connector Locations
8
Audio Output:
Line:
Unbalanced - RCA jacks, 2VRMS/+6dBV max (digital
clipping), with an impedance of 470 ohms and a load
impedance of > 10 k ohms
Balanced - Nine pin D connector, -16 dBm maximum, with
an impedance of 47 ohms and a load impedance of 600
There are 5 connectors located on the SX-9/SX -11 audio board's
metal mounting bracket. See Figure 5, 7 & 8. for more information
on the SX -9/SX-11 connectors.
Audio Output:
Line:
Balanced - Nine pin D connector 0 to +26dBu(digital
clipping) - software selectable with 1dBu resolution, with an
impedance of 50 ohms and a load impedance of 600 ohms
Unbalanced - 1/8” stereo mini jack 2VRMS/+6dBV max
(digital clipping), with an impedance of 470 ohms and a
load impedance of 10k ohms
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