VC
MS
switch
(N.O.)
+
Rear
Mic
Zero Biased
Receiver
+
Front
Mic
+
-
CS44
T-coil
Figure 3: Assembly diagram
INTRODUCTION
The GB3212 hybrid comprises a highly versatile, advanced
digital signal processing system.
Further IIR filter blocks process the front microphone and
rear microphone signals. Two biquad filters, "miccomp1"
and "miccomp2", are used to match the rear microphone's
gain and phase to that of the front microphone. After the
miccomp filters, more filters are used to provide an
adjustable group delay to create the desired polar response
pattern during the calibration process.
Configuration data stored in non-volatile memory defines
hearing aid parameters. This data needs to be uploaded to
the hybrid before the circuit becomes functional. The
GB3212 hybrid is programmed via the SDA pin using
industry-standard programming boxes.
In the Telecoil mode gains are trimmed during Cal/Config
Configuration data is generated by an ARK product
component library (DLL). Like Gennum's other digital
products, the GB3212 is fully supported by Gennum's
software tools available from the Gennum ARK web site
http://ark.gennum.com.
process
to
compensate
for
microphone/telecoil
mismatches.
The FRONTWAVE® block is followed by two cascaded
biquad filters, "pre1" and "pre2". These filters can be used
for frequency response shaping before the signal goes from
the CODEC chip into the DSP chip. When FRONTWAVE® is
not enabled, the miccomp filters can be used for frequency
response shaping also.
SIGNAL PATH
There are two main audio input signal paths. The first path
contains the Front Microphone and second path contains
the Rear Microphone or Telecoil input as selected by a
programmable MUX. The front microphone input is
intended as the main Microphone audio input for single
microphone applications. In FRONTWAVE® operation, a
multimicrophone signal is used to produce a directional
hearing instrument response. The two audio inputs are
buffered, sampled and converted into digital form using
dual A/D converters. The digital outputs are converted into
a 32kHz, 20-bit digital audio signal.
After passing through the biquad filters the signal enters the
DSP chip. At this point, the signal is converted to 16kHz
and 16-bit. The DSP chip runs the following signal
processing algorithms:
•
•
•
•
•
•
frequency analysis
4 channel WDRC
16 band frequency shaping
16 band noise reduction
frequency band synthesis
adaptive feedback cancellation
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