SB3231-E1-T Datasheet SB3231-E1-T, SB3231-E1 | P7-P9

RHYTHM SB3231

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RHYTHM SB3231 OVERVIEW

Rhythm SB3231 is a DSP system implemented on

ON Semiconductor’s WOLVERINEt hardware platform.

Wolverine is the hearing industry’s first 90 nm

Silicon−on−Chip platform enabling design of

highly−efficient and flexible hearing aid solutions. The

device is packaged for easy integration into a wide range of

applications from CIC to BTE. Rhythm SB3231 can be used

as a programmable or trimmer adjustable device. It may be

configured as one, two or four channels with linear or

WDRC processing. Configuration data stored in

non−volatile memory defines hearing−aid parameters.

Rhythm SB3231 can be programmed via the SDA or I

programming interfaces.

The DSP core implements Adaptive Feedback

Cancellation, Adaptive Noise Reduction, FrontWave

directionality, compression, wideband gain, and volume

control. The Adaptive Feedback Canceller reduces acoustic

feedback while offering robust performance against pure

tones.

The Rhythm SB3231 contains a 256 kbit EEPROM and

can be used for both programmable and trimmer based

applications. It is compatible with ON Semiconductor’s

ARK tools and SOUNDFIT fitting software.

During trimmer mode operation, a low−speed A/D circuit

monitors the positions of up to four manual trimmers and

a VC potentiometer. Trimmer position changes are

immediately interpreted and translated into new circuit

parameter values, which are then used to update the signal

path.

FUNCTIONAL BLOCK DESCRIPTION

A/D and D/A Converter

The system’s A/D converter is a 2nd−order sigma−delta

modulator operating at a 2.048 MHz sample rate.

The system’s input is pre−conditioned with anti−alias

filtering and a programmable gain pre−amplifier. The

analog output is oversampled and modulated to produce

a 1−bit pulse density modulated (PDM) data stream. The

digital PDM data is then decimated down to pulse−code

modulated (PCM) digital words at the system’s sampling

rate of 32 kHz.

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 64−times

oversampled, 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.

Analog Inputs

Rhythm SB3231 provides for up to four analog inputs,

Microphone 1 (MIC1), Microphone 2 (MIC2), Telecoil

(TCOIL) and Direct Audio Input (DAI) with the following

configurable front end modes:

• 1 Mic Omni

• 1 Mic Omni (Rear channel only)

• FrontWave Directional

• 2 Mic Omni (MIC1 + MIC2)

• DAI

• TCOIL

• 1 Mic Omni + TCOIL

• 1 Mic Omni + DAI

Attenuation can be applied to the input when mixing with

either TCOIL or DAI inputs.

Analog input signals should be ground referenced to

MGND. (Microphones, telecoils, DAI). MGND is

internally connected to GND to minimize noise, and should

not be connected to any external ground point.

Channel Processing

Figure 5 represents the I/O characteristic of independent

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

four main regions:

• Low input level expansion (squelch) region

• Low input level linear region

• Compression region

• High input level linear region (return to linear)

Figure 5. 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

Channel I/O processing is specified by the Squelch

threshold (SQUELCHTH) and any four of the following

five parameters (only four of the five properties are

independent):

• Low level gain (LLGAIN)

• Lower threshold (LTH)

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• High level gain (HLGAIN)

• Upper threshold (UTH)

• Compression ratio (CR)

During the Parameter Map creation, constraints are

applied to the compression parameters to ensure that the I/O

characteristics are continuous. Parameter adjustments

support two popular styles of compression ratio adjustment:

• The compression region of the I/O curve pivots about

the upper threshold. As the compression ratio trimmer

is adjusted, high−level gain remains constant while the

low−level gain changes.

• The compression region of the I/O curve pivots about

the lower threshold. Low−level gain remains constant

as the compression ratio trimmer is adjusted.

The squelch region within each channel implements a low

level noise reduction scheme (1:3 expansion) for listener

comfort. This scheme operates in quiet listening

environments (programmable threshold) to reduce the gain

at very low levels.

Automatic Telecoil

The automatic telecoil feature in Rhythm SB3231 is to be

used with memory D programmed with the telecoil or

MIC + TCOIL front end configuration. The feature enables

the part to transition to memory D upon the closing of

a switch connected to MS2. With the feature enabled and

a reed switch connected to MS2, the static magnetic field of

a telephone handset will close the switch whenever the

handset is brought close to the device, causing the hybrid to

change to memory D. The part will transition back to the

initial memory once the switch is deemed opened after

proper debouncing.

A debounce algorithm with a programmable debounce

period is used to prevent needless switching in and out of

memory D due to physical switch bounces when MS2 is

configured for automatic telecoil. Upon detecting a close to

open switch transition, the debounce algorithm monitors the

switch status. The debounce algorithm switches the device

out of memory D only once the switch signal has been

continuously sampled open over the specified debounce

period.

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, Rhythm

SB3231’s AFC does not reduce the hearing aid’s gain. The

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

The third−generation AFC (see Figure 6) allows for an

increase in the stable gain

of the hearing instrument while

minimizing artefacts for music and tonal input signals. As

with previous products, the feedback canceller provides

completely automatic operation.

1. Added stable gain will vary based on hearing aid style and

acoustic setup. Please refer to the Adaptive Feedback

Cancellation Information note for more details.

Figure 6. 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 64−band filter bank (32 bands at 16 kHz

sampling) 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 64 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

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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.

FrontWave Directional Microphones

The FrontWave feature is implemented in two front−end

modes on Rhythm SB3231: static directional and

two−microphone omnidirectional. Both these front−end

modes are designed to operate using two closely spaced

omnidirectional microphones connected to the VIN1 and

VIN2 inputs.

In static directional mode, FrontWave synthesizes

a directional response pattern by delaying the

rear−microphone signal and subtracting it from the front

microphone signal. Various microphone response patterns

can be obtained by adjusting the rear−microphone time

delay.

In two−microphone omnidirectional mode, FrontWave

synthesizes a secondary omnidirectional response pattern

by delaying the front microphone signal and adding it to the

rear microphone signal. The resulting omnidirectional

microphone signal possesses a noise floor that is

approximately 3 dB lower than that provided by a single

microphone (assuming both microphones have similar noise

floors).

The FrontWave feature includes three parameters that can

be set via external software: time delay, rear−microphone

compensation filter and a low−frequency boost filter

intended for static directional mode. Time delay can be

configured using IDS software. It determines the polar

patter in static directional mode and accounts for

microphone spacing in two−microphone omnidirectional

mode. The rear−microphone compensation filter provides

a means to adjust the rear−microphone sensitivity so that it

can better match the front microphone. It is controlled

automatically through Cal/Config software. The

low−frequency boost filter compensates for the 6 dB/octave

roll−off in frequency response that occurs in directional

mode. The amount of low frequency equalization is

programmable through IDS.

NOTE: For optimum FrontWave operation,

ON Semiconductor recommends using matched

microphone pairs.

The time delay implemented using FrontWave is not

explicitly limited within the system. Optimum accuracy is

obtained, however, for smaller time delays. For example, in

32 kHz operation, a time delay of 81.5 microseconds can be

achieved with a maximum deviation of 5% over a bandwidth

of 0 to 4 kHz. This allows a microphone port spacing of

approximately 28 mm. For 16 kHz operation, a similar

accuracy is observed for a time delay of 78.1 microseconds,

corresponding to a port spacing of approximately 26.8 mm.

Smaller time delays can be implemented with improved

accuracy.

Volume Control, Trimmers and Switches

External Volume Control

The volume of the device can either be set statically via

software or controlled externally via a physical interface.

Rhythm SB3231 supports both analog and digital volume

control functionality, although only one can be enabled at

a time. Digital control is supported with either a momentary

switch or a rocker switch. In the latter case, the rocker switch

can also be used to control memory selects.

Analog Volume Control

Both the external (analog) volume control and trimmers

work with a three−terminal 100 kW − 360 kW variable

resistor. The volume control can have either a log or linear

taper, which is selectable via IDS. It is possible to use a VC

with up to 1 MW of resistance, but this could result in a slight

decrease in the resolution of the taper.

Trimmers

The trimmer interface provides the ability to control up to

19 hearing aid parameters through up to four trimmers.

A single trimmer parameter can have up to 16 values and

a single trimmer can control multiple parameters (e.g.,

Trimmer 1 can control compression ratio in all four channels

simultaneously). The trimmer must be three−terminal

100 kW to 360 kW variable resistors and have a linear taper.

Parameters that can be assigned to trimmers include Noise

Reduction, Low Cut, High Cut, Compression Ratio,

Wideband Gain, Tinnitus Noise Level, Crossover

Frequency, Lower Threshold, Upper Threshold, EQ Gain,

Squelch Threshold, High Level Gain, Low Level Gain,

AGC−O Threshold, Static Volume Control and Peak Clipper

Threshold.

NOTE: There may be limitations to which parameters

can be used together.

Digital Volume Control

The digital volume control makes use of two pins for

volume control adjustment, VC and D_VC, with

momentary switches connected to each. Closure of the

switch to the VC pin indicates a gain increase while closure

to the D_VC pin indicates a gain decrease. Figure 7 shows

how to wire the digital volume control to Rhythm SB3231.

Figure 7. Wiring for Digital Volume Control

D_VC

GND

P1-P3 P4-P6 P7-P9 P10-P12 P13-P15 P16-P18

SB3231-E1-T

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Manufacturer:

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Description:

Audio DSPs PRECONFIG DSP: RHYTH

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