English
Language : 

SA3291 Datasheet, PDF (11/20 Pages) ON Semiconductor – Preconfigured Wireless DSP System for Hearing Aids
AYRE SA3291
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 (ADM)
ON Semiconductor’s Adaptive Directional Microphone
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
reduce current consumption. While in this omni−directional
state, the algorithm will periodically check for conditions
warranting the enabling of the adaptive directional
microphone.
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.
In−Situ Datalogging − iLog 4.0
The Ayre SA3291 has a datalogging function that records
information every 4 s to 60 minutes (programmable) about
the state of the hearing aid and its environment to
non−volatile memory. The function can be enabled with the
ARK software and information collection will begin the
next time the hybrid is powered up. This information is
recorded over time and can be downloaded for analysis.
The following parameters are sampled:
• Battery level
• Volume control setting
• Program memory selection
• Environment
• Ambient sound level
• Length of time the hearing aid was powered on
• Wireless audio and phone streaming
The information is recorded using two methods in parallel:
• Short−term method − a circular buffer is serially filled
with entries that record the state of the first five of the
above variables at the configured time interval.
• Long−term method − increments a counter based on the
memory state at the same time interval as that of the
short−term method. Based on the value stored in the
counter, length of time the hearing aid was powered on
can be calculated.
www.onsemi.com
11