Implementation in FPAA:

Field programmable analog arrays can be used to implement flexible subwoofer

conditioner designs such as those comprising a subsonic filter, audio compressor,
Linkwitz Transform equalizer and crossover filter

Each of the CAMs has function-specific parameters from which the design tools
determine the underlying circuit structures and settings, and map these onto the
configurable analog blocks inside the FPAA. The development tools then deliver
configuration information that must be loaded into the FPAA at power-up, either directly
from a PROM (for those designs that are configured at time of manufacture) or — more
significantly — from a microcontroller that accompanies the FPAA. The latter case
allows features of the design to change in the field under software control.
Dynamic and automatic tuning
To allow in situ tuning, the microcontroller that accompanies the FPAA is supplied with
simple software instructions to modify any specific aspect of the design. These
instructions are custom-generated according to the form of the data or C-code
application programming interface that has been identified by the FPAA development
tools in which the circuit is designed.
From the user's point of view, this means that an active subwoofer system can be
adjusted in the field using some familiar mechanism — such as an infrared remote
control — rather than requiring an audio technician to adjust a filter notch in the back of
a speaker cabinet.
Taking this concept just one step further is the ability of these systems to self-tune. For
example, the peak response of speakers within the low-frequency spectrum is always a
function of the environment in which they are placed. With an FPAA implementation, the
system microcontroller can instruct the FPAA to stimulate the speaker with a low-
frequency signal and monitor ambient sound level from a microphone. As the
microcontroller programs the FPAA to deliver a sweep of oscillator frequencies, it can
capture the return characteristic to determine which frequencies need suppression.
Having performed this calibration sequence, the FPAA is assigned its primary role as a
subwoofer conditioner circuit, and the newly determined frequency compensation
characteristic can be applied to the Linkwitz Transform circuit.
Band Pass Biquadratic Filter

Two different circuits can realize the band pass version of the biquadratic filter.  The
implementation that best fits the combination of CAM Parameter values is selected. 
The Roman numeral in the upper right corner of the CAM symbol identifies the selected
circuit

Peak detector

This CAM creates a peak detector with programmable decay rate.  Positive or negative
peaks can be detected.  This CAM has continuous output that is always valid.

Pole zero CAM

This CAM should be used with a held input in order to achieve the closest match to the
stated transfer function.

Gain Voltage Controlled

This CAM implements a stage with variable gain, controlled by the input voltage at a
second input.  The profile of gain versus Control voltage is a user specified function with
256 quantization steps.  If the “Output Hold” option is off, the output has amplifier input
offset compensation during phase 1.  If it is on, the output will be held through the entire
clock period without offset compensation.  This CAM has restrictions on the “Control”
input connection.

Low pass filter

This CAM creates a full cycle, bilinear (single pole) filter, in a low pass, high pass, all
pass or pole and zero configuration. The high pass, all pass and pole and zero versions
should be used with a held input signal. The filter has programmable corner frequency
and pass band gain.

Subwoofer :–

A type of loudspeaker designed specifically to reproduce very low pitch sounds, usually
those under 150 Hz. Customarily seen as large cabinets located on the floor, with one
or two 15 or 18-inch low-frequency drivers, although it has been proven by some
industry pioneers that size does not always matter. Although there are very few
naturally-occurring sounds that require a frequency response below 40 Hz, special
effects in both theatre and film often make good use of the subwoofer.
 Working Principle:

As an alternative to discrete component and DSP solutions, these field programmable

analog arrays (FPAAs) give audio designers the flexibility they need to develop
subwoofer designs that deliver pleasing results whatever the contours of the
environment in which they are used.
Where a classic discrete design is used, the two most commonly seen parts of any
active subwoofer system are the low-pass crossover filter that attunes the subwoofer to
other speakers in the system, balancing their respective frequency responses, and the
speaker amplifier itself. Additional signal conditioning is frequently required, however.
Subsonic filters are often used to remove near-dc signals, which can damage the driver
coils. But one side effect of these filters is the attenuation of low-frequency tones (15 to
50 Hz), which are often highly desirable to give that special "punch" to the audio
experience. Conversely, we see the accentuation of resonant frequencies of the
speaker housing or environment, which need suppressing.

 Equalizers:

Equalizers are units that affect certain portions of the frequency spectrum with a
boost (gain) or cut (attenuation). From your electrical engineering knowledge,
you know that a circuit that affects a certain portion of a given frequency
spectrum is called a filter. You will see the word "filter" used quite often when
describing bands of equalization. We'll go into more detail later.

 Reverberation:

Reverberation is defined as the phenomenon that occurs when sound waves are
reflected and reinforced; this phenomenon occurs naturally in nearly all enclosed
spaces. Certain types of signal processors are designed to artificially create the
effect of reverberation; originally designed to enhance sound recordings, making
the recording sound "fuller," or "more live," it is also widely found in live sound
systems to enhance the sound of a particular hall, or for creating other-worldly
effects.
 Echo/Delay:

Delay and Echo should not be confused with Reverberation. While reverberation
is the homogeneous sound image caused by multiple reflections, delay refers to
one or more distinct sound images, repeated over time. Delay units allow for
special, other-worldly effects when used in a recording situation; in live sound
delay units are used to correct time differences between different systems of
loudspeakers. (More will be discussed later).

 Compressors and Limiters:

Compressors and Limiters are signal processors that reduce the dynamic

range of an audio signal. Essentially, a compressor and a limiter are the same
thing; the difference in nomenclature merely comes from the actual, practical use
of the unit. The compressor is designed to prevent audio signals from exceeding
a given level-- in simplest terms, it keeps things from getting too loud. Limiters do
the same thing, but more drastically. Television and broadcast systems use a lot
of compression in order to fit their program material into a specified bandwidth; in
practical live sound compressors can be used on vitriolic keyboards or
unpredictable voices; many loudspeaker processors have a form of built-in limiter
to protect the loudspeaker itself from damaging volume levels.

 Noise Gates:

A Noise Gate is a signal processor that turns off or significantly attenuates the
audio signal when the signal level falls below a certain threshold. In practical
application, desired program will pass through the unit unaltered, but low-level
hiss and noise audible between sections of desired program will not be allowed
passage. Noise Gates are used in recording situations; for instance when micing
a drumkit a noise gate can be inserted on the snare-drum microphone and
adjusted such that when other drums are played, the noise gates prevents the
signal from passing, whilst when the snare-drum is played, the signal, which
would be a higher volume level than the other drums as picked up from the
snare-drum microphone, would be allowed passage. In live sound systems, noise
gates are sometimes used as a sort of "automatic mute." Many corporate
installations use noise-gates to automatically mute microphones that are not in
use, which contributes to higher gain before feedback and a lower noise floor.
 Amplifiers:

Amplifiers are a basic building block of electrical engineering. In its theoretical

form, an amplifier takes an electrical signal and increases its power in a linear
fashion- that is to say, the output waveform matches the input waveform, only it's
more powerful. While the linearity of common power amplifiers is not necessarily
perfect, that is essentially what a common power amplifier does. Power
amplifiers are units designed to take an audio signal and make it powerful
enough to drive a loudspeaker; due to the loudspeaker construction and its
inherent inefficiency, a lot of power is required to produce sound. Other types of
amplifiers used in audio include preamplifiers, which are small amplifiers that
take an audio signal from a microphone and increase its power so that it can be
better manipulated.