SA3291A-E1-T Datasheet SA3291A-E1, SA3291A-E1-T | P10-P12

AYRE SA3291

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environments and make the necessary adjustments to the

parameters in the audio path, such as ADM, ANR, WDRC,

FBC, in order to optimize the hearing aid settings for the

specific environment.

iSceneDetect will gradually make the adjustments so the

change in settings based on the environment is smooth and

virtually unnoticeable. This feature will enable the hearing

aid wearer to have an instrument which will work in any

environment with a single “memory”.

EVOKE Advanced Acoustic Indicators

Advanced acoustic indicators provide alerting sounds that

are more complex, more pleasing and potentially more

meaningful to the end user than the simple tones used on

previous products. The feature is capable of providing

pulsed, multi−frequency pure tones with smooth on and off

transitions and also damped, multi−frequency tones that can

simulate musical notes or chords.

A unique indicator sound can be assigned to each of the

ten system events: memory select (A, B, C, D, E or F), low

battery warning, digital VC movement and digital VC

minimum/maximum. Each sound can consist of a number of

either pure tones or damped tones but not both.

A pure tone sound can consist of up to four tones, each

with a separate frequency, amplitude, duration and start

time. Each frequency component is smoothly faded in and

out with a fade time of 64 ms. The start time indicates the

beginning of the fade in. The duration includes the initial

fade−in period. By manipulating the frequencies, start times,

durations and amplitudes various types of sounds can be

obtained (e.g., various signalling tones in the public

switched telephone network).

A damped tone sound can consist of up to six tones, each

with a separate frequency, amplitude, duration, start time

and decay time. Each frequency component starts with

a sudden onset and then decays according to the specified

time constant. This gives the audible impression of a chime

or ring. By manipulating the frequencies, start times,

durations, decays and amplitudes, various musical melodies

can be obtained.

Acoustic indication can be used without the need to

completely fade out the audio path. For example, the

low−battery indicator can be played out and the user can still

hear an attenuated version of the conversation.

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. The forward path of the hearing aid is not

affected. Unlike adaptive notch filter approaches, the Ayre

SA3291’s AFC does not reduce the hearing aid’s gain. The

AFC is based on a time−domain model of the feedback path.

The Ayre SA3291 third−generation AFC provides an

increase in added stable gain and minimal artefacts for music

and tonal input signals. As with previous products, the

feedback canceller in the Ayre SA3291 provides completely

automatic operation. The feedback canceller can be

activated in any front−end mode except for Telecoil−only or

DAI−only mode.

When the AFC is enabled, it is highly recommended that

you either have all channels with Squelch ON or all channels

with Squelch OFF. If you choose to have all channels with

Squelch ON then there is an additional requirement to have

all Squelch thresholds above the microphone noise floor. If

you require any assistance in determining what threshold

levels to set, please contact the applications department at

ON Semiconductor. Squelch ON/OFF does not incur any

current penalty. When Squelch and AFC are both ON, the

Squelch is limited to 1:2 expansion.

Figure 5. Adaptive Feedback Canceller (AFC)

Block Diagram

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

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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.

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There are 750 log entries plus 6 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, VC, 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 Ayre SA3291 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 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 its 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 Ayre SA3291 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.

HRX Head Room Expander

The Ayre SA3291 has an enhanced Head Room Expander

(HRX) circuit that increases the input dynamic range of the

Ayre SA3291 without any audible artifacts. This is

accomplished by dynamically adjusting the pre−amplifier’s

gain and the post−A/D attenuation depending on the input

level.

Channel Processing

Figure 6 represents the I/O characteristic of independent

AGC channel processing. The I/O curve can be divided into

the following main regions:

• Low input level expansion (squelch) region

• Low input level linear region

• Compression region

• High input level linear region (return to linear)

Figure 6. Independent Channel I/O Curve Flexibility

−100

−90

−80

−70

−60

−50

−40

−30

−20

−10

−120 −110 −100 −90 −80 −70 −60 −50 −40 −30 −20

OUTPUT LEVEL (dBV)

INPUT LEVEL (dBV)

Low Level

Gain

Compression

Ratio

High Level

Gain

Squelch

Threshold

Lower

Threshold

Upper

Threshold

The I/O characteristic of the channel processing can be

adjusted in the following ways:

• Squelch threshold (SQUELCHTH)

• Low level gain (LLGAIN)

• Lower threshold (LTH)

• High level gain (HLGAIN)

• Upper threshold (UTH)

• Compression ratio (CR)

To ensure that the I/O characteristics are continuous, it is

necessary to limit adjustment to a maximum of four of the

last five parameters. During Parameter Map creation, it is

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