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CS6422 Datasheet(PDF) 32 Page - Cirrus Logic

Part # CS6422
Description  Enhanced Full-Duplex Speakerphone IC
PDF  48 Pages
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Manufacturer  CIRRUS [Cirrus Logic]
Direct Link  http://www.cirrus.com
Logo CIRRUS - Cirrus Logic

CS6422 Datasheet(HTML) 32 Page - Cirrus Logic

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CS6422
32
4. DESIGN CONSIDERATIONS
When designing the CS6422 into a system, it is im-
portant to keep several considerations in mind.
These concerns can be loosely grouped into three
categories: algorithmic considerations, circuit de-
sign considerations, and system design consider-
ations.
4.1
Algorithmic Considerations
The CS6422 facilitates full-duplex hands-free
communication via many algorithms running on
the Digital Signal Processor that is the core of the
CS6422. Among these are the algorithms that per-
form the adaptive filtering, the half-duplex switch-
ing, digital volume control, and supplementary
echo suppression.
4.1.1
Full-Duplex Mode
Full-duplex hands-free communication is achieved
through a technique called adaptive filtering. The
basic principle behind adaptive filtering is that the
acoustic path between speaker and microphone can
be modeled by a transfer function which can be dy-
namically determined by an adaptive digital filter.
This principle assumes good update control and
speech/tone detection algorithms to prevent the fil-
ter from mistraining.
4.1.1.1
Theory of Operation
Figure 10 illustrates how the adaptive filter can
cancel echo and reduce loop gain. The echo path of
the system is between points B and C: the speaker
to microphone coupling. A signal injected at A
(sometimes called a “training signal”) is sent both
to B, the input of the echo path, and to F, the input
of the adaptive filter. The signal at B is modified by
the acoustic transducers (speaker and microphone)
and the environment, and received at point C (as an
“Echo”). Meanwhile, assume that the adaptive fil-
ter has exactly the right transfer function to match
the echo path BC, and so the signal at point D is ap-
proximately equal to the signal at point C. After
these are subtracted by the summing element, all
that is left is the error signal at point E, which
should be very small.
If a person were to speak into the microphone at
point C, that signal would pass through the sum-
ming element unchanged because the adaptive fil-
ter had no comparable input to subtract out. In this
manner, the person at A and the person at C may si-
multaneously speak and A will not hear his own
echo.
In the real world, the echo path is not static. It will
change, for example, when people move in the
Σ
Adaptive Filter
A
E
D
B
C
+
-
F
Figure 10. Simplified Acoustic Echo Canceller Block Diagram
CS6422
32
DS295F1



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