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SA3286-E1 Datasheet(PDF) 8 Page - ON Semiconductor

No. de pieza SA3286-E1
Descripción Electrónicos  Pre-configured DSP System
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Fabricante Electrónico  ONSEMI [ON Semiconductor]
Página de inicio  http://www.onsemi.com
Logo ONSEMI - ON Semiconductor

SA3286-E1 Datasheet(HTML) 8 Page - ON Semiconductor

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INSPIRIA SA3286
www.onsemi.com
8
Figure 5. Adaptive Feedback Canceller (AFC)
Block Diagram
G
Σ
H’
H
+
Feedback path
Estimated feedback
Feedback Path Measurement Tool
The Feedback Path Measurement Tool uses the onboard
feedback cancellation algorithm and noise generator to
measure the acoustic feedback path of the device. The noise
generator is used to create an acoustic output signal from the
hearing aid, some of which leaks back to the microphone via
the feedback path. The feedback canceller algorithm
automatically calculates the feedback path impulse response
by analyzing the input and output signals. Following a
suitable adaptation period, the feedback canceller
coefficients can be read out of the device and used as an
estimate of the feedback−path impulse response.
Adaptive Noise Reduction
The noise reduction algorithm is built upon a high
resolution 128−band filter bank enabling precise removal of
noise. The algorithm monitors the signal and noise activities
in these bands, and imposes a carefully calculated
attenuation gain independently in each of the 128 bands.
The noise reduction gain applied to a given band is
determined by a combination of three factors:
Signal−to−Noise Ratio (SNR)
Masking threshold
Dynamics of the SNR per band
The SNR in each band determines the maximum amount
of attenuation to be applied to the band − the poorer the SNR,
the greater the amount of attenuation. Simultaneously, in
each band, the masking threshold variations resulting from
the energy in other adjacent bands is taken into account.
Finally, the noise reduction gain is also adjusted to take
advantage of the natural masking of ‘noisy’ bands by speech
bands over time.
Based on this approach, only enough attenuation is
applied to bring the energy in each ‘noisy’ band to just below
the masking threshold. This prevents excessive amounts of
attenuation from being applied and thereby reduces
unwanted artifacts and audio distortion. The Noise
Reduction algorithm efficiently removes a wide variety of
types of noise, while retaining natural speech quality and
level. The level of noise reduction (aggressiveness) is
configurable to 3, 6, 9 and 12 dB of reduction.
Directional Microphones
In any directional mode, the circuitry includes a fixed
filter for compensating the sensitivity and frequency
response differences between microphones. The filter
parameters are adjusted during product calibration.
A dedicated biquad filter following the directional block
has been allocated for low frequency equalization to
compensate for the 6 dB/octave roll−off in frequency
response that occurs in directional mode. The amount of low
frequency equalization that is applied is programmable.
ON Semiconductor recommends using matched
microphones. The maximum spacing between the front and
rear microphones cannot exceed 20 mm (0.787 in).
Adaptive Directional Microphones
ON Semiconductor’s Adaptive Directional Microphone
(ADM) algorithm is a two−microphone processing scheme
for hearing aids. It is designed to automatically reduce the
level of sound sources that originate from behind or the side
of the hearing−aid wearer without affecting sounds from the
front. The algorithm accomplishes this by adjusting the null
in the microphone polar pattern to minimize the noise level
at the output of the ADM. The discrimination between
desired signal and noise is based entirely on the direction of
arrival with respect to the hearing aid: sounds from the front
hemisphere are passed unattenuated whereas sounds
arriving from the rear hemisphere are reduced.
The angular location of the null in the microphone polar
pattern is continuously variable over a range of 90 to 180
degrees where 0 degrees represents the front.
The location of the null in the microphone pattern is
influenced by the nature of the acoustic signals (spectral
content, direction of arrival) as well as the acoustical
characteristics of the room. The ADM algorithm steers a
single, broadband null to a location that minimizes the
output noise power. If a specific noise signal has frequency
components that are dominant, then these will have a larger
influence on the null location than a weaker signal at a
different location. In addition, the position of the null is
affected by acoustic reflections. The presence of an acoustic
reflection may cause a noise source to appear as if it
originates at a location other than the true location. In this
case, the ADM algorithm chooses a compromise null
location that minimizes the level of noise at the ADM
output.
Automatic Adaptive Directional Microphones
When Automatic ADM mode is selected, the adaptive
directional microphone remains enabled as long as the
ambient sound level is above a specific threshold and the
directional microphone has not converged to an
omni−directional polar pattern. On the other hand, if the
ambient sound level is below a specific threshold, or if the
directional
microphone
has
converged
to
an
omni−directional polar pattern, then the algorithm will
switch to single microphone, omni−directional state to


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