GA3219 Datasheet, PDF (6/14 Pages) Gennum Corporation – Venture-TM Digital DSP System

VENTURE GA3219
Front
Mic
+
T−coil
MS
switch
(N.O.)
Rear
Mic
+
+
−
Zero Biased
Receiver
CS44
VC
Figure 4. Typical Hearing Instrument Assembly Diagram
VENTURE GA3219 OVERVIEW
Venture GA3219 is a DSP system with adaptive
algorithms that run on an advanced hardware platform. This
hardware platform is a combination of a DSP core and a high
fidelity audio CODEC. As well, thinSTAX packaging
provides easy integration into a wide range of applications
from CIC to BTE.
The audio functions implemented on the CODEC include
tone generation, peak clipping and cross fading between
audio paths. The DSP core implements Frontwave
directional processing, programmable filters, adaptive
algorithms, compression, wideband gain, and volume
control. The adaptive algorithms include Adaptive Noise
Reduction and Adaptive Feedback Cancellation.
The Adaptive Noise Reduction reduces audible noise in a
low distortion manner while preserving perceived speech
levels. The Adaptive Feedback Canceller reduces acoustic
feedback while offering robust performance against pure
tones. As well, Venture GA3219 contains security features
to protect clients’ Intellectual Property against device
cloning and software piracy.
Venture GA3219 utilizes the power and capabilities of the
hardware platform to deliver advanced features and
enhanced performance previously unavailable to a product
in its class.
This data sheet is part of a documentation set available for
this product.
SIGNAL PATH
There are two main audio input signal paths. The first path
contains the front microphone and the second path contains
the rear microphone, telecoil or direct audio 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
multi−microphone 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
32 kHz or 16 kHz, 20−bit digital audio signal.
Further IIR filter blocks process the front microphone and
rear microphone signals. One biquad filter is used to match
the rear microphone’s gain to that of the front microphone.
After that, other filtering is used to provide an adjustable
group delay to create the desired polar response pattern
during the calibration process.
In the Telecoil mode gains are trimmed during Cal/Config
process to compensate for microphone/telecoil mismatches.
The FrontWave block is followed by four cascaded biquad
filters: pre1, pre2, pre3 and pre4. These filters can be used
for frequency response shaping before the signal goes
through channel and adaptive processing.
The channel and adaptive processing consists of the
following:
• Frequency band analysis
• 1, 2 or 4 channel WDRC
• Eight logarithmically spaced band frequency shaping
(graphic EQ)
• 128 frequency band adaptive noise reduction
• Frequency band synthesis
• Phase cancellation adaptive feedback reduction
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