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NetPro Certification Courseware for NetPro Certified Systems Engineer – N.C.S.

SOUND BOARDS
Sound is an area of the PC that has been largely overlooked in early systems. Aside
from a simple, oscillator-driven speaker, the early PCs were mute. Driven largely by
the demand for better PC games, designers developed stand-alone soundboards that
could read sound data recorded in separate files, then reconstruct those files into basic
sound, music and speech.

Since the beginning of the decade, those early sound boards have blossomed into an
array of powerful, high-fidelity sound products, capable of duplicating voice,
orchestral soundtracks, and real-life sounds with uncanny realism . Not only have
sound products helped the game industry to mature, but they have been instrumental
in the development of multimedia technology (the integration of sound and picture),
as well as Internet Web phones and other communication tools.

THE RECORDING PROCESS

All sound starts as pressure variations traveling through the air. Sound can come from
almost anywhere—a barking dog, a laughing child, a fire engine’s siren, a person
speaking. You get the idea. The process of recording sound to a hard drive requires
sound to be carried through several manipulations. First, sound must be translated
from pressure variations in the air to analog electrical signals. A microphone
accomplishes this. These analog signals are amplified by the sound card, then
digitized (converted to a series of representative digital words each taken at a fixed
time interval). The resulting stream of data is processed and organized through the use
of software, which places the data (as well as any overhead or housekeeping data) into
a standard file format. The file is saved to the drive of choice (typically, a hard drive).
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THE PLAYBACK PROCESS

Simply speaking, the playback process is virtually the reverse of recording. A


software application opens a sound file on the hard drive, and then passes the digital
data back to the sound card. Data is translated back into equivalent analog levels—
ideally, the reconstructed shape of the analog signal closely mimics the original
digitized signal. The analog signal is amplified, and then passed to a speaker. If the
sound was recorded in stereo, the data is divided into two channels that are separately
converted back to analog signals, amplified, and sent to their corresponding speakers.
Speakers convert the analog signal back into traveling pressure waves that you can
hear.
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THE CONCEPT OF “SAMPLING”

To appreciate the intricacies of a sound card’s operation, you must understand the
concept of digitization (otherwise known as sampling). In principle, sampling is a
very straight forward concept; an analog signal is measured periodically, and its
voltage at each point in time is converted to a digital number. The device that
performs this conversion is known as an Analog-to-Digital Converter (ADC). It
sounds simple enough in principle, but it has some important wrinkles. The problem
with sampling is that a digitizer circuit has to capture enough points of an analog
waveform to reproduce it faithfully. The example in figure illustrates the importance
of sampling rate. Waveforms A and B represent the same original signal. Waveform
A is sampled at a relatively slow rate—only a few samples are taken. The problem
comes when the signal is reconstructed with a Digital-to-Analog Converter (DAC).
NetPro Certification Courseware for NetPro Certified Systems Engineer – N.C.S.E

As you see, there are not enough sample points to reconstruct the original signal. As a
result, some of the information in the original signal is lost. This form of distortion is
known as aliasing. Waveform B is the same signal, but it is sampled at a much higher
rate. When that data is reconstructed, the resulting signal is a much more faithful
reproduction of the original. As a rule, a signal should be sampled at least twice as
fast as the highest frequency contained in the signal—this is known as Nyquist’s
Sampling Theorem. The lowest standard sampling rate used with today’s soundboards
is 11kHz—this allows fair reproduction of normal speech and vocalization (up to
about 5.5kHz). However, most low-end soundboards can digitize signals up to 22kHz.
Unfortunately, the human range of hearing is about 22kHz. To capture sounds
reasonably well throughout the entire range of hearing, you would need a sampling
rate of 44kHz—often known as CD-quality sampling because it is the same rate used
to record audio on CDs. The disadvantage to high sampling rates is disk space (and
sound file size). Each sample is a piece of data, so the more samples taken each
second, the larger and faster a file grows.

DATA BITS VS. SOUND QUALITY

Not only does the number of samples affect sound quality, but also the precision (or
number of bits) of each sample. Suppose that each sample is converted to a 4-bit
number. That means a number can represent each point from 0 to 15—not much
precision there. If 8 bits are used for each sample, 256 discrete levels can be
supported. But the most popular configuration is 16-bit conversion, which allows a
sample to be represented by one of 65,536 levels. At that level of resolution, samples
form a very close replica of the original signal. Many of today’s soundboards are 16-
bit.

THE ROLE OF MIDI

Although the majority of a sound card is geared toward handling the recording and
playback of sound files, the Musical Instrument Digital Interface (MIDI) port has
become an inexpensive and popular addition to many sound-card designs. The MIDI
standard is defined by hardware, software, and electrical interconnections. At the core
of a MIDI interface is a synthesizer IC. Unlike a sound file, which basically contains
the digital equivalent of an analog waveform, a MIDI file is a set of instructions for
playing musical notes. Each note is sent to the synthesizer, along with duration, pitch,
and timing specifications. The synthesizer can be made to replicate a variety of
musical instruments, such as a piano, guitar, harmonica, flute—you name it. The
high-end soundboards are capable of synthesizing a small orchestra. Because most
synthesizers can process several channels simultaneously, the MIDI standard supports
playing a number of “instruments” (or voices) at the same time. Thus, very high-
quality music can be produced with MIDI on a PC. The two most common
synthesizer types are FM and Wavetable. Pre-recorded MIDI files can be read from a
NetPro Certification Courseware for NetPro Certified Systems Engineer – N.C.S.E

storage device, such as a hard-drive file, or from CD-ROM (many games include an
orchestral-quality MIDI soundtrack on the CD). The MIDI data is passed through to
the soundboard’s synthesizer which reproduces the sound, and out to the amplified
speakers. If you plan to compose music yourself, you can interface a MIDI instrument
to the soundboard’s MIDI port. Using MIDI sequencer software, the notes played on
the instrument will be heard through the speaker, as well as recorded to the MIDI file
on the hard drive. Notice that you do not need a MIDI instrument to playback a MIDI
file, but you need instrument and sequencer software to create a MIDI file. Also,
because MIDI is not sound (but rather sound “blueprints”), the same MIDI
composition entered on a keyboard can be played back as a harp, or a guitar, or a
flute.
NetPro Certification Courseware for NetPro Certified Systems Engineer – N.C.S.E

INSIDE A SOUND BOARD

Now that you are aware of the major functions a soundboard must perform, you can
see those functions in the context of a complete board. Figure shows a simplified
block diagram of a soundboard. It is important that your own particular soundboard
might differ somewhat, but all contemporary boards should contain these subsections.
The core element of a soundboard is the Digital Signal Processor (DSP). A DSP is a
variation of a microprocessor that is specially designed to manipulate large volumes
of digital data. Like all processor components, the DSP requires memory. A ROM
contains all of the instructions needed to operate the DSP and direct the board’s major
operations. A small quantity of RAM serves two purposes: it provides a “scratch pad”
area for the DSP’s calculations and it serves as a buffer for data traveling to or from
the PC bus. Signals entering the soundboard are passed through an amplifier stage and
provided to an A/D converter. When recording occurs, the DSP runs the A/D
converter and accepts the resulting conversions for processing and storage. Signals
delivered by a microphone are typically quite faint, so they are amplified
significantly. Signals delivered to the “line” input are often much stronger (such as the
output from a CD player or stereo preamp), so it receives less amplification. For
signals leaving the soundboard, the first (and often most important) stop is the mixer.
NetPro Certification Courseware for NetPro Certified Systems Engineer – N.C.S.E

The mixer combines CD-audio, DSP sound output, and synthesizer output into a
single analog channel. Because most soundboards now operate in a stereo mode, most
have two mixer channels and amplifier stages. The audio amplifier stage(s) boost the
analogs signal for delivery to stereo speakers. If the sound will be driving a stereo
system, a “line” output provides a separate output. Amplifier output can be adjusted
by a single master volume control located on the rear of the board. Finally, a MIDI
controller is provided to accommodate the interface of a MIDI instrument to the
soundboard. In many cases, the interface can be jumpered to switch the controller to
serve as a joystick port. That way, the soundboard can support a single joystick if the
MIDI instrument will not be used. MIDI information processed by the DSP will be
output to the on-board synthesizer.

Using Microphones

An ever-growing number of sound card owners are using their sound cards to record
sound or broadcast sound over the Internet through such applications as Web Phone.
Sound recording demands the use of microphones, and not all microphones work
properly with every soundboard. Often, the user mistakes a poor microphone response
as being a problem with the sound card. This part of the chapter looks at some
important considerations for choosing and using a microphone.

MICROPHONE TYPES

The three types of microphones are: dynamic, condenser, and electrets condenser. All
three-microphone types are available for soundboards:

 Dynamic: Dynamic microphones are typically hand-held or desktop units.


They have a larger response range and usually sound better than condenser
microphones. A dynamic microphone does not require phantom power
because the diaphragm element in the microphone can create enough electric
current for the soundboard to use.

 Condenser: Condenser microphones are the small multimedia microphones


that are typically sold with computers. When you open a new soundboard and
take the microphone out of the box, it is almost always a condenser
microphone. They do not have as good a response range as dynamic
microphones, and they also have a smaller diaphragm— this demands
phantom power from the soundboard.
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 Electrets: condenser Electrets condenser microphones are basically condenser


microphones with a built-in battery for power. They have the same response as
a condenser microphone, but they do not require phantom power to operate.
Some electrets condenser microphones will allow you to remove this internal
battery. With the battery not installed, phantom power would be required.

PHANTOM POWER

The next question is “What is phantom power?” Phantom power is simply a small,
low-current power supply on the soundboard, which is used to power some
microphones. Such devices as dynamic microphones can produce enough current on
their own to avoid the use of phantom power, but condenser microphones demand
phantom power as a current source. Here’s the main problem with today’s
soundboards—not all of them provide switchable phantom power. Ideally,
soundboards (such as the Ensoniq Soundscape) would provide phantom power and
allow you to jumper the phantom power on or off, depending on which microphone
you plan to use. If you use a dynamic microphone, you’d switch phantom power off.
If you use a condenser microphone, you’d switch phantom power on. If a soundboard
does not provide phantom power at all, you’re stuck using a dynamic microphone or a
powered electrets condenser microphone. If a soundboard provides full-time phantom
power (and you cannot turn it off), you’ll need to stay with a condenser microphone.
You can probably see the potential for trouble here. If you use a condenser
microphone on an un-powered soundboard, the microphone will not work at all (or
generate little more than faint noise). On the other hand, plugging a dynamic or
electrets microphone into a powered sound board will usually result in severe
clipping—once again, you’ll capture little more than noise.

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