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參數(shù)資料
| 型號(hào)： | GB3212 |
| 廠商： | Gennum Corporation |
| 英文描述： | DUET⑩ DIGITAL Advanced DSP System with FRONTWAVE㈢ |
| 中文描述： | 二重奏⑩先進(jìn)的DSP數(shù)字系統(tǒng)的FRONTWAVE㈢ |
| 文件頁數(shù)： | 6/10頁 |
| 文件大小： | 118K |
| 代理商： | GB3212 |
GENNUM CORPORATION
20352 - 2
6 of 10
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Once the signal has been processed by the DSP chip it
goes back into the CODEC chip. On the CODEC chip there
are four more cascaded biquad filters — "post1", "post2",
"post3" and "post4". These biquad filters are followed by the
tone generator, AGCo block and two more biquad filters —
"postagco1" and "postagco2". The last stage is in the signal
path is the D/A H-bridge.
FUNCTIONAL BLOCK DESCRIPTION
ADAPTIVE FEEDBACK CANCELLER
The adaptive feedback canceller (AFC) reduces acoustic
feedback by forming an estimate of the hearing aid
feedback signal and then subtracting this estimate from the
hearing aid input. Therefore the forward path of the hearing
is not affected. Unlike adaptive notch filter approaches,
DUET's AFC does not reduce the hearing aid's gain. The
AFC is based on a time-domain model of the feedback
path.
Figure 4: Adaptive Feedback Canceller (AFC) block diagram
ADAPTIVE NOISE REDUCTION
Noise reduction is applied independently in each of 16
frequency bands. The algorithm utilizes perceptual criteria
to determine the audibility of individual noise bands. More
attenuation is applied to those bands where the noise is
most audible. Less attenuation is applied where the noise is
inaudible. This maximizes the perceptual benefit of noise
reduction and also reduces the audible artifacts that are
often associated with adaptive noise reduction algorithms.
The attenuation applied to a given band is determined by a
combination of two factors: the SNR and the masking
threshold. The SNR estimate in each band determines the
maximum amount of attenuation that will be applied to that
band (the poorer the SNR the greater the amount of
attenuation). At the same time the masking threshold
resulting from the energy in adjacent bands is also
estimated. Only enough attenuation is applied to bring the
energy in each 'noise' band to just below the masking
threshold. This prevents excessive amounts of attenuation
from being applied and thereby reduces unwanted artifacts
and distortion.
A/D AND D/A CONVERTERS
The system's two A/D converters are 2nd-order sigma-delta
modulators, which operate at a 2.048MHz sample rate. The
system's two audio inputs are pre-conditioned with antialias
filtering and programmable gain pre-amplifiers. These
analog outputs are over sampled and modulated to
produce two, 1-bit pulse density modulated (PDM) data
streams. The digital PDM data is then decimated down to
pulse-code modulated (PCM) digital words at the system
sampling rate of 32kHz.
The D/A is comprised of a digital, 3rd-order sigma-delta
modulator and an H-bridge. The modulator accepts PCM
audio data from the DSP path and converts it into a 32-
times over-sampled, 1-bit PDM data stream, which is then
supplied to the H-bridge. The H-bridge is a specialized
CMOS output driver used to convert the 1-bit data stream
into a low-impedance, differential output voltage waveform
suitable for driving zero-biased hearing aid receivers.
HRX HEAD ROOM EXPANDER
The GB3212 has an enhanced Head Room Expander
(HRX) circuit, which increases the input dynamic range of
the DUET DIGITAL without any unwanted audible
artifacts. This is accomplished by dynamically adjusting the
preamplifier's gain and the post-A/D attenuation depending
on the input level.
FRONTWAVE DIRECTIONALITY
The FRONTWAVE block provides the resources necessary
to implement directional microphone processing. The block
accepts inputs from both a front and rear microphone and
provides a synthesized directional microphone signal as its
output. The directional microphone output is obtained by
delaying the rear microphone signal and subtracting it from
the front microphone signal. Various microphone response
patterns can be obtained by adjusting the time delay.
The FRONTWAVE circuit also provides a fixed filter for
compensating the sensitivity and frequency response
differences between microphones. The filter parameters are
adjusted during product calibration.
One of the generic IIR filters following the FRONTWAVE
block ("pre1") has been allocated for low frequency
equalization to compensate for the 6dB/octave roll off in
frequency response that occurs in directional mode. The
amount of low frequency equalization that is applied can be
determined during product calibration.
Gennum recommends using matched microphones with
FRONTWAVE, although calibration is fully possible using
unmatched
microphones.
unmatched microphones will result in no difference in
directionality. However, over a longer period of time
unmatched microphones are more likely to drift apart and
result in poor directional characteristics.
Initially,
calibration
using
G
Σ
H'
H
+
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Feedback path
Estimated feedback
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| GB3221 | Power PARAGON-TM DIGITAL Four Channel DSP System with FRONTWAVE-R |
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相關(guān)代理商/技術(shù)參數(shù) |
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| GB3215 | 制造商:GENNUM 制造商全稱:GENNUM 功能描述:PARAGON⑩ DIGITAL Four Channel DSP System with FRONTWAVE |
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| GB3221 | 制造商:GENNUM 制造商全稱:GENNUM 功能描述:Power PARAGON-TM DIGITAL Four Channel DSP System with FRONTWAVE-R |
| GB3225 | 制造商:GENNUM 制造商全稱:GENNUM 功能描述:PARAGON⑩ DIGITAL Four Channel DSP System with FRONTWAVE |
| GB-323 | 制造商:LUCKY-LIGHT 制造商全稱:LUCKY-LIGHT 功能描述:Round Type Low Profile LED Lamps (4.7mm) |
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