SA3286 Datasheet, PDF (9/16 Pages) ON Semiconductor – Pre-configured DSP System

INSPIRIA SA3286
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 2.0
The Inspiria SA3286 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
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.
There are 750 log entries plus 4 memory select counters
which are all protected using a checksum verification. A
new log entry is made whenever there is a change in memory
state, volume control, or battery level state. A new log entry
can also be optionally made when the environmental sound
level changes more than the programmed threshold, thus it
is possible to log only significantly large changes in the
environmental level, or not log them at all.
The ARK software iLog graph displays the iLog data
graphically in a way that can be interpreted to counsel the
user and fine tune the fitting. This iLog graph can be easily
incorporated into other applications or the underlying data
can be accessed to be used in a custom display of the
information.
Tinnitus Treatment
The Inspiria SA3286 has an internal white noise generator
that can be used for Tinnitus Treatment. The noise can be
attenuated to a level that will either mask or draw attenuation
away from the user’s tinnitus. The noise can also be shaped
using low−pass and/or high−pass filters with adjustable
slopes and corner frequencies.
As shown in Figure 1, the Tinnitus Treatment noise can be
injected into the signal path either before or after the volume
control (VC) or it can be disabled. If the noise is injected
before the VC then the level of the noise will change along
with the rest of the audio through the device when the VC is
adjusted. If the noise is injected after the VC then it is not
affected by VC changes.
The Tinnitus Treatment noise can be used on it’s own
without the main audio path in a very low power mode by
selecting the Tinnitus Treatment noise only. This is
beneficial either when amplification is not needed at all by
a user or if the user would benefit from having the noise
supplied to them during times when they do not need
acoustic cues but their sub−conscious is still active, such as
when they are asleep.
The ARK software has a Tinnitus Treatment tool that can
be used to explore the noise shaping options of this feature.
This tool can also be easily incorporated into another
software application.
Narrow−band Noise Stimulus
The Inspiria SA3286 is capable of producing Narrow−
band Noise Stimuli that can be used for in situ audiometry.
Each narrow−band noise is centred on an audiometric
frequency. The duration of the stimuli is adjustable and the
level of the stimuli are individually adjustable.
A/D and D/A Converters
The system’s two A/D converters are second order
sigma−delta modulators operating at a 2.048 MHz 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 32 kHz.
The D/A is comprised of a digital, third order sigma−delta
modulator and an H−bridge. The modulator accepts PCM
audio data from the DSP path and converts it into a 64−times
or 128−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.
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