SA3291

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© Semiconductor Components Industries, LLC, 2016 April, 2016 − Rev. 8

1 Publication Order Number:

Preconfigured Wireless DSP System for Hearing Aids

Description

Ayre/C0116 SA3291 is a preconfigured wireless DSP system utilizing Near−Field Magnetic Induction (NFMI) technology. Ayre SA3291 enables hearing aids to wirelessly synchronize program modes and volume control and stream telecoil signals from one hearing aid to the other. When coupled with a relay device, Ayre SA3291 enables features such as stereo audio streaming via Bluetooth ® in addition to remote control functionality. Featuring iSceneDetect /C0116 environmental classification, adaptive noise reduction, superior feedback cancellation, fully automated and adaptive microphone directionality, and up to 8−channel WDRC, Ayre SA3291 is ideal for high−end, full−featured wireless products. Binaural Synchronization − The binaural synchronization feature allows two hearing aids to wirelessly synchronize adjustments such as program modes or volume control. By working together as one system, user control is simplified. Binaural T elecoil − The binaural telecoil wirelessly streams telecoil audio signals from one hearing aid to the other. This enables hearing aid users to hear phone calls in both ears, improving speech intelligibility. Stereo Audio Streaming − Stereo audio signals can be streamed wirelessly from a relay device to hearing aids equipped with Ayre SA3291. A relay device can use Bluetooth or other far−field wireless technology to wirelessly connect with TVs, music players, mobile phones or other audio sources. Acoustic Environment Classification − The iSceneDetect 1.0 environmental classification algorithm is capable of analyzing the hearing aid wearer’s acoustic environment and automatically optimizes the hearing aid to maximize comfort and audibility. iLog/C0093 4.0 Datalogging − Enables the recording of various hearing aid parameters such as program selection, volume setting and ambient sound levels. The sampling interval can be configured to record from every 4 seconds up to once every 60 minutes. The fitting system can present the data to help the fitting specialist fine tune the hearing aid and counsel the wearer. EVOKE /C0091 Advanced Acoustic Indicators − Allows manufacturers to provide more pleasing, multi−frequency tones simulating musical notes or chords to indicate events such as program or volume changes. Automatic Adaptive Directionality − The automatic Adaptive Directional Microphone (ADM) algorithm automatically reduces the level of sound sources that originate from behind or to the side of the hearing aid wearer without affecting sounds from the front. The algorithm can also gather input from the acoustic environment and automatically select whether directionality is needed or not, translating into additional current savings. Adaptive Feedback Cancellation − Automatically reduces acoustic feedback. It allows for an increase in the stable gain while minimizing artifacts for music and tonal input signals. www.onsemi.com SIP32 HYBRID CASE 127DW PAD CONNECTION MARKING DIAGRAM SA3291A−E1 XXXXXX SA3291A = Specific Device Code E1 = RoHS Compliant Hybrid XXXXXX = Work Order Number (Bottom View) GND VIN1 PGND 9101112 TIN VREG VC GPIO7 DIGVC NGND NVB VBP DAI VB GPIO0/ MS1 GPIO1/ MS2 PCSDA PCCLK OUT− OUT+ GPIO4 N/CGPIO2GPIO3 N/CN/CN/CN/CN/CN/C MGND VIN2 See detailed ordering and shipping information on page 17 of this data sheet.

ORDERING INFORMATION

www.onsemi.com Adaptive Noise Reduction − The Ayre SA3291 adaptive noise reduction algorithm monitors noise levels independently in 128 individual bands and employs advanced psychoacoustic models to provide user comfort. Tinnitus Masking − The Ayre SA3291 is equipped with a noise source that can be used to mask tinnitus. The noise can be shaped and attenuated and then summed into the audio path either before or after the volume control. In−situ T one Generator − The narrow−band noise stimulus feature can be used for in−situ validation of the hearing aid fitting. The frequency, level and duration of the stimuli are individually adjustable. Other Key Features – Ayre SA3291 also supports the following features: FRONTWA VE ® directional processing, built−in feedback path measurement, cross fading between audio paths for click−free program changes, 16−band graphic equalizer, 8 generic biquad filters (configurable as parametric or other filter types), programming speed nhancements, optional peak clipping, flexible ompression adjustments, direct interfaces to analog or digital volume control, rocker switch, direct audio input and telecoil. Ayre SA3291 also encompasses industry−leading security features to avoid cloning and software piracy.

Features

• Wireless: ♦ Complete NFMI Wireless Transceiver of Bi−directional Transmission ♦ Channel Rate of 298 kbps ♦ Wireless Data Transfer ♦ CROS/BiCROS ♦ Relay Device Reference Design Available
• Advanced Research Algorithms: ♦ iSceneDetect Environmental Classification ♦ Automatic Adaptive Directional Microphones (ADM) ♦ FrontWave Directionality ♦ 128−band Adaptive Noise Reduction ♦ Adaptive Feedback Cancellation (AFC)
• Streaming Stereo and Mono Audio
• iLog 4.0 Datalogging
• Tinnitus Masking Noise Generator
• Evoke Acoustic Indicators
• Auto Telecoil with Programmable Delay
• 1, 2, 4, 6 or 8 Channel WDRC
• Feedback Path Measurement Tool
• AGC−O with Variable Threshold, Time Constants, and Optional Adaptive Release
• 16−band Graphic Equalizer
• Narrow−band Noise Stimulus
• SDA or I2C Programming
• 8 Biquadratic Filters
• 4 Analog Inputs
• 16 kHz or 8 kHz Bandwidth
• 6 Fully Configurable Memories with Audible Memory Change Indicator
• 96 dB Input Dynamic Range with HRX Headroom Extension
• 28−bit Fingerprint Security System and Other Security Features to Protect Against Device Cloning and Software Policy
• High Fidelity Audio CODEC
• Soft acoustic Fade Between Memory Changes
• Drives Zero−bias 2−terminal Receivers
• Internal or External Digital V olume Control with Programmable Range
• Rocker Switch Support
• 20−bit Audio Processing
• thinSTAX® Packaging
• E1 RoHS Compliant Hybrid thinSTAX Packaging Hybrid typical dimensions: 0.250 x 0.145 x 0.065 in. (6.35 x 3.68 x 1.65 mm) 1. Nominal dimensions of Hybrid only; height of solder bump not included.

Figure 1. Hybrid Block Diagram

Table 1. ABSOLUTE MAXIMUM RATINGS should not be assumed, damage may occur and reliability may be affected. WARNING: Electrostatic Sensitive Device − Do not open packages or handle except at a static−free workstation. WARNING: Moisture Sensitive Device − RoHS Compliant; Level 4 MSL. Do not open packages except under controlled conditions. Table 2. ELECTRICAL CHARACTERISTICS (Supply Voltage VB = NVB = 1.25 V; Temperature = 25°C)

Table 2. ELECTRICAL CHARACTERISTICS (Supply Voltage VB = NVB = 1.25 V; Temperature = 25°C) (continued)

100 Hz − 8 kHz

performance may not be indicated by the Electrical Characteristics if operated under different conditions.

Table 3. I2C TIMING period, the first clock pulse is generated.

1. A device must internally provide a hold time of at least 300 ns for the PC_SDA signal to bridge the undefined region of the falling edge of PC_CLK.
2. The maximum t HD;DAT has only to be met if the device does not stretch the LOW period (tLOW) of the PC_CLK signal.
3. A Fast−mode I 2C−bus device can be used in a Standard−mode I2C−bus system, but the requirement tSU;DAT P250ns must then be met.

to the Standard−mode I2C−bus specification) before the PC_CLK line is released.

1. C b = total capacitance of one bus line in pF.

Figure 2. I2C Mode Timing

www.onsemi.com SIGNAL PATH The integrated NFMI transceiver is designed to provide many highly desirable features in a wireless hearing aid product. With minimal processing overhead, the transceiver and its built−in MAC and link control function provides networked mode support for 3 or more devices, unidirectional low latency stereo audio streaming and high−speed data transfer from one transmitter device to two hearing instruments. The use of NFMI technology avoids all the propagation problems that exist in an RF−based wireless system. While having a small wireless range typically limited to less than 60 cm, depending on the transmitter device, the NFMI system has good immunity to the human body blockage effect and it is not sensitive to RF interference. The very high Q−factor for its coil antenna ensures a very small operating system bandwidth in the radio front end. There are two main audio input signal paths. The first path contains the front microphone and the second path contains the rear microphone, telecoil or direct audio input as selected by a programmable MUX. The front microphone input is intended as the main microphone audio input for single microphone applications. In iSceneDetect, FrontWave, ADM or Automatic ADM operation, a multi−microphone signal is used to produce a directional hearing instrument response. The two audio inputs are buffered, sampled and converted into digital form using dual A/D converters. The digital outputs are converted into a 32kHz or 16 kHz, 20−bit digital audio signal. Further IIR filter blocks process the front microphone and rear microphone signals. One biquad filter is used to match the rear microphone’s gain to that of the front microphone. After that, other filtering is used to provide an adjustable group delay to create the desired polar response pattern during the calibration process. In iSceneDetect, ADM and Automatic ADM, the two microphone inputs are combined in an adaptive way while in FrontWave operation the combination is static. In the Telecoil mode gains are trimmed during Cal/Config process to compensate for microphone/telecoil mismatches. The FrontWave block is followed by four cascaded biquad filters: pre1, pre2, pre3 and pre4. These filters can be used for frequency response shaping before the signal goes through channel and adaptive processing. The channel and adaptive processing consists of the following:

• Frequency band analysis
• 1, 2, 4, 6 or 8 channel WDRC
• 16 frequency shaping bands (spaced linearly at 500 Hz intervals, except for first and last bands)
• 128 frequency band adaptive noise reduction
• Frequency band synthesis After the processing the signal goes through two more biquad filters, post1 and post2, which are followed by the AGC−O block. The AGC−O block incorporates the Wideband Gain and the volume control (VC). There are also two more biquad filters, post3 and post4, and the Peak Clipper. The last stage in the signal path is the D/A H−bridge. White noise can be shaped, attenuated and then added into the signal path at two possible locations: before the VC (between the Wideband Gain and the VC) or after the VC (between post 4 and the Peak Clipper) as shown in Figure 1. FUNCTIONAL BLOCK DESCRIPTION Binaural Synchronization The binaural synchronization feature allows two hearing aids to wirelessly synchronize adjustments such as program modes or volume control. Binaural Telecoil The binaural telecoil wirelessly streams telecoil audio signals from one hearing aid to the other, enabling hearing aid users to hear phone calls in both ears. In this mode, the user can also adjust the VC on the ear not using the telephone. If the hearing aids become disconnected by going out of range, in 94% of cases they will reconnect in less than a second. Wireless Streaming Stereo audio signals can be streamed wirelessly from a relay device to hearing aids equipped with Ayre SA3291. A relay device can use Bluetooth or other far−field wireless technology to wirelessly connect with TVs, music players, mobile phones or other audio sources. Streaming is primarily a one way mode, with the transmitter streaming audio or commands to all devices within its range. Streaming devices can operate across a greater distance due to the larger transmit antenna and greater transmit power. A receiver must be configured in IDS to allow streaming audio or remote control commands to be received. CROS/BiCROS To compensate for single sided deafness, AYRE SA3291 can be configured for use as a CROS/BiCROS device. It does this by transmitting audio from a microphone on a device on the completely deaf ear to the device on the aided or unaided ear over a wireless link. iSceneDetect 1.0 Environment Classification The iSceneDetect feature, when enabled, will sense the environment and automatically control the enhancement algorithms without any user involvement. It will detect speech in quiet, speech in noise, wind, music, quiet and noise

environment with a single “memory”. simulate musical notes or chords. either pure tones or damped tones but not both. switched telephone network). hear an attenuated version of the conversation. AFC is based on a time−domain model of the feedback path. Squelch is limited to 1:2 expansion. Figure 5. Adaptive Feedback Canceller (AFC) estimate of the feedback−path impulse response. attenuation gain independently in each of the 128 bands.

• 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

www.onsemi.com 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
• V olume 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 necessary to select four parameters as user adjustable, or fixed, and to allow one parameter to be calculated. The squelch region within each channel implements a low level noise reduction scheme (1:2 or 1:3 expansion ratio) for listener comfort. This scheme operates in quiet listening environments (programmable threshold) to reduce the gain at very low levels. When the Squelch and AFC are both enabled it is highly recommended that the Squelch be turned on in all channels and that the Squelch thresholds be set above the microphone noise floor (see Adaptive Feedback Canceller). The number of compression channels is programmable in ARKonline ® and can be 1, 2, 4, 6 or 8. Telecoil Path The telecoil input is calibrated during the Cal/Config process. To compensate for the telecoil/microphone frequency response mismatch, a first order filter with 500 Hz corner frequency is implemented. Through ARKonline, it is possible to implement a telecoil compensation filter with an adjustable corner frequency. To accommodate for the gain mismatch, the telecoil gain is adjusted to match the microphone gain at 500 Hz or 1 kHz (default) and is selectable in ARKonline. There is also a telecoil gain adjustment parameter that can be enabled in ARKonline and set in IDS, enabling manual adjustment of the telecoil gain compensation. Automatic Telecoil The Ayre SA3291 is equipped with an automatic telecoil feature, which causes the hybrid to switch to a specific memory upon the closing of a switch connected to MS2. This feature is useful when MS2 is connected to a switch, such as a reed switch, that is open or closed depending on the presence of a static magnetic field. Memory D can be programmed to be the telecoil or mic+telecoil memory so that, when a telephone handset is brought close to such a switch, its static magnetic field closes the switch and causes the hybrid to change to memory D. However, it is possible that the hearing aid wearer may move his or her head away from the telephone handset momentarily, in which case it is undesirable to immediately change out of telecoil mode and then back in moments later. The A yre SA3291 has a debounce circuit that prevents this needless switching. The debounce circuit delays the device from switching out of memory D when MS2 is configured as a static switch in ‘D−only’ mode. The debounce time is programmable to be 1.5, 3.5 or 5.5 seconds after the switch opens (i.e., the handset is moved away from the hearing instrument) or this feature can be disabled. DAI Path The DAI input can be adjusted using a first order filter with a variable corner frequency similar to the telecoil compensation filter. Through ARKonline, it is possible to implement this DAI filter to set either a static or adjustable corner frequency. The Mic plus DAI mode mixes the Mic1 and DAI signals. The Mic1 input signal is attenuated by 0, −6 or −12 dB before being added to the DAI input signal. The DAI input also has gain adjustment in 1 dB steps to assist in matching it to the Mic1 input level. Graphic Equalizer The Ayre SA3291 has a 16−band graphic equalizer. The bands are spaced linearly at 500 Hz intervals, except for the first and the last band, and each one provides up to 24 dB of gain adjustment in 1 dB increments. Biquad Filters Additional frequency shaping can be achieved by configuring generic biquad filters. The transfer function for each of the biquad filters is as follows: H(z) /C0043b0 /C0041b1 /C0032z−1 /C0041b2 /C0032z−2 1 /C0041a1 /C0032z−1 /C0041a2 /C0032z−2 Note that the a0 coefficient is hard−wired to always be ‘1’. The coefficients are each 16 bits in length and include one sign bit, one bit to the left of the decimal point, and 14 bits to the right of the decimal point. Thus, before quantization, the floating−point coefficients must be in the range −2.0 ≤ x < 2.0 and quantized with the function: round /C0466x /C0032214/C0467 After designing a filter, the quantized coefficients can be entered into the PreBiquads or PostBiquads tab in the Interactive Data Sheet. The coefficients b0, b1, b2, a1, and a2 are as defined in the transfer function above. The parameters meta0 and meta1 do not have any effect on the signal processing, but can be used to store additional information related to the associated biquad. The underlying code in the product components automatically checks all of the filters in the system for stability (i.e., the poles have to be within the unit circle) before updating the graphs on the screen or programming the coefficients into the hybrid. If the Interactive Data Sheet receives an exception from the underlying stability checking code, it automatically disables the biquad being modified and display a warning message. When the filter is made stable again, it can be re−enabled. Also note that in some configurations, some of these filters may be used by the product component for microphone/telecoil compensation, low−frequency EQ, etc. If this is the case, the coefficients entered by the user into IDS are ignored and the filter designed by the software is programmed instead. Rocker Switch The A yre SA3291 is equipped with a rocker switch feature that can perform both VC adjustments or an audio memory switch.

• Digital VC
• Momentary Memory Select
• Mixed Mode In Mixed Mode, the switches behaviour is configurable to be set to that a short or long press of the switch will invoke either a memory or VC change (i.e., a short press is a memory select, a long press is a VC change). There is a programmable threshold that can be used to set the timing behaviour. Volume Control The VC can be either external (digital VC) or programmable. When using a Digital V olume Control (DVC) with the Ayre SA3291, the switch should be connected to the VC and D_VC pins 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 DVC to SA3291. A toggle switch can be used as a DVC, momentarily connecting the VC to either Vreg or GND. By connecting the VC to Vreg, the volume will be increased one step, and by connecting the VC to GND, the volume will be decreased one step. The following parameters can be programmed into the hybrid to specify the DVC functionality:
• DVC enable or disable
• V olume up/down step size of 1 dB, 2 dB, 3 dB or 4 dB
• V olume up/down beep frequency and volume
• DVC range between 0 dB and 48 dB in 1 dB steps
• Default DVC value when the hybrid is powered up
• V olume up/down beep enable
• Max/Min beep enable
• Max/Min beep frequency & volume If the Max/Min beep is enabled then when the volume has been incremented to the maximum value of the specified DVC range the device will play two beeps to indicate that it cannot increase the volume any more. The same is true for decrementing the volume and reaching the minimum value of the DVC range.

Figure 7. Wiring for Digital Volume Control tremendous flexibility in switching between configurations. select options are selectable via the settings tab in IDS. valid memory, a button press causes memory A to be loaded. parameter to ‘Momentary’ and ‘Donly’ to ‘disabled’. part loads in the last select memory. parameter to ‘Momentary’ and ‘Donly’ to ‘enabled’.

Table 4. DYNAMIC EXAMPLE WITH FOUR VALID PROGRAM MODES (T = momentary switch is toggled; 0 = OPEN; 1 = HIGH) depending on the state of the switches. invalid, the part defaults to memory A. parameter to ‘static’ and ‘Donly’ to ‘disabled’. Table 5. MEMORY SELECTED IN STATIC SWITCH ON memory is selected depending on the state of the switches. parameter to ‘static’ and ‘Donly’ to ‘enabled’. Table 6. MEMORY SELECTED IN STATIC SWITCH ON memory before reaching the final memory. invalid, the part defaults to memory A. on or off, and during SDA programming. power on reset delay function.

www.onsemi.com Power On Reset Delay The programmable POR delay controls the amount of time between power being connected to the hybrid and the audio output being enabled. This gives the user time to properly insert the hearing aid before the audio starts, avoiding the temporary feedback that can occur while the device is being inserted. During the delay period, momentary button presses are ignored. NOTE: The values set in IDS are relative values from 0 to 11 seconds; not absolute. The POR delay is relative to the configuration loaded on the WOLVERINE platform. Power Management Functionality As the voltage on the hearing aid battery decreases, an audible warning is given to the user indicating the battery life is low. In addition to this audible warning, the hearing aid takes other steps to ensure proper operation given the weak supply. The exact hearing aid behaviour in low supply conditions depends on the selected POR mode. The hearing aid has three POR modes:

• Shallow Reset Mode
• Deep Reset Mode
• Advanced Mode Shallow Reset Mode In Shallow Reset mode, the hearing aid will operate normally when the battery is above 0.95 V . Once the supply voltage drops below 0.95 V the audio will be muted and remain in that state until the supply voltage rises above 1.1 V. Once the supply voltage drops below the control logic ramp down voltage, the device will undergo a hardware reset. At this point, the device will remain off until the supply voltage returns to 1.1 V . When the supply voltage is below the control logic voltage, but above 0.6 V and rises above the

1.1 V turn on threshold, the device will activate its output

and operate from the memory that was active prior to reset. If the supply voltage drops below 0.6 V , and rises above the

1.1 V turn on threshold, the device will reinitialize, activate

its output and operate from memory A. Deep Reset Mode In Deep Reset mode, the hearing aid will operate normally when the battery is above 0.95 V . Once the supply voltage drops below 0.95 V the audio will be muted. The device remains in this state until the supply voltage drops below the hardware reset voltage of 0.6 V . When this occurs, the device will load memory A and operate normally after the supply voltage goes above 1.1 V . Advanced Reset Mode Advanced Reset Mode on Ayre SA3291 is a more sophisticated power management scheme than shallow and deep reset modes. This mode attempts to maximize the device’s usable battery life by reducing the gain to stabilize the supply based on the instantaneous and average supply voltage levels. Instantaneous supply fluctuations below

0.95 V can trigger up to two 3 dB, instantaneous gain

reductions. Average supply drops below 0.95 V can trigger up to eighteen, 1 dB average gain reductions. While operating with no instantaneous gain reductions, an instantaneous supply voltage fluctuation below 0.95 V will trigger an immediate 3 dB gain reduction. A waiting period of 30 seconds is in place after the first instantaneous gain reduction. Only after the waiting period has elapsed will an instantaneous supply voltage fluctuation trigger the second 3 dB gain reduction. While an instantaneous gain reduction is being applied, the instantaneous supply voltage level will be checked every 30 seconds and a 3 dB gain reduction removed should the level be above a certain threshold. Should the average supply voltage drop below 0.95 V , the device will begin to reduce the gain by 1 dB every 10 seconds until either the average supply voltage rises above

0.95 V or all 18 average gain reductions have been applied,

at which point the audio path will be muted. If the average supply voltage returns to a level above 1.1 V , the audio path will first be un−muted, if required. The gain will then be increased by 1 dB every 10 seconds until either the average supply voltage drops below 1.1 V , or all average gain reductions have been removed. No action is taken while the average supply voltage resides between 0.95 V and 1.1 V . NOTE: Instantaneous and average gain reductions are adjusted independently. When the instantaneous voltage falls below the hardware shutdown voltage, the device will undergo a hardware reset. When it turns back on because the voltage has risen above the turn−on threshold, it will behave the same as it would in shallow reset mode. Low Battery Notification Notification of the low battery condition via an acoustic indicator is optionally performed when the battery voltage drops below a configurable low battery notification threshold. The low battery indicator is repeated every five minutes until the device shuts down. Software and Security The Ayre SA3291 incorporates the following security features to protect the device from cloning and against software piracy:

• DLL protection by password − prevents a third party from using IDS to reconfigure parts.
• Hybrid authentication by 128−bit fingerprint to identify parts in application software − prevents a third party from cloning a device’s EEPROM because the fingerprint cannot be overwritten. Special functions can be used in fitting software to reject parts that do not match the expected fingerprint. This would prevent the piracy of fitting software.
• DLL to hybrid pairing by using a software key in ARK to match product libraries with client software − a part can be ‘locked’ at manufacturing time so that it only communicates with the library it was programmed with.

www.onsemi.com This prevents a third party from potentially upgrading a device with a different library in IDS or other application software. Full software support is provided for every stage of development from design to manufacturing to fitting. For details, refer to the Getting Started with the ARK Software information note. SDA and I2C Communication The Ayre SA3291 can be programmed using the SDA or I2C protocol. During parameter changes, the main audio signal path of the hybrid is temporarily muted using the memory switch fader to avoid the generation of disturbing audio transients. Once the changes are complete, the main audio path is reactivated. Any changes made during programming are lost at power−off unless they are explicitly burned to EEPROM memory. Improvements have been made to the ARK software for the Ayre SA3291 resulting in increased communication speed. Certain parameters in ARKonline can be selected to reduce the number of pages that need to be read out. In SDA mode, the Ayre SA3291 is programmed via the SDA pin using industry standard programming boxes. I mode is a two wire interface which uses the SDA pin for bidirectional data and CLK as the interface clock input. I programming support is available on the HiPro (serial or USB versions) and ON Semiconductor’s DSP Programmer 3.0. Power Supply Considerations The Ayre SA3291 was designed to accommodate high power applications. AC ripple on the supply can cause instantaneous reduction of the battery’s voltage, potentially disrupting the circuit’s function. The Ayre SA3291 hybrids have a separate power supply and ground connections for the output stage. This enables hearing instrument designers to accommodate external RC filters to minimize any AC ripple from the supply line. Reducing this AC ripple greatly improves the stability of the circuit and prevents unwanted reset of the circuit caused by spikes on the supply line. For more information on properly designing a filter to reduce supply ripple, refer to the Using DSP Hybrids in High Power Applications Initial Design Tips information note (AND9028). Material Handling It is recommended that any exposed material be dry baked for a minimum of 37 hours ±1 hour at 90 °C ±5°C before storage again. For general handling specifications, please refer to the ’How to Store, Reflow and Solder ON Semiconductor Hybrids’ application note (AND8493). Device Package Shipping† SA3291A−E1 32 Pad Hybrid 25 Units / Bubble Pack SA3291A−E1−T 32 Pad Hybrid 250 Units / Tape & Reel †For information on tape and reel specifications, including part orientation and tape sizes, please refer to our Tape and Reel Packaging Specifications Brochure, BRD8011/D. Hybrid Jig Ordering Information To order a Hybrid Jig Kit for Ayre SA3291 contact your Sales Account Manager or FAE and use part number SA3410GEVK. Kit includes: − Two Ayre SA3291 Hybrid Jig Evaluation Boards − Two NFMI Antennas − Two Ayre SA3291 Hybrids − Two Programming Cables To order a Hybrid Jig Board for Ayre SA3291 contact your Sales Account Manager or FAE and use part number SA3410GEVB.

Table 7. PAD POSITION AND DIMENSIONS

www.onsemi.com PACKAGE DIMENSIONS SIP32 6.35x3.68 CASE 127DW ISSUE O SEATING PLANE NOTES: 1. DIMENSIONING AND TOLERANCING PER ASME Y14.5M, 1994. 2. CONTROLLING DIMENSION: MILLIMETERS. 3. COPLANARITY APPLIES TO SPHERICAL CROWNS OF SOLDER BUMPS. 4. DIMENSIONS b, b1, L AND L2 ARE MEASURED AT THE MAXIMUM BUMP DIMENSION PARALLEL TO DATUM C. THE POSITIONAL TOLERANCE APPLIES TO ALL OF THE SOLDER BUMPS. DIM A MIN MAX −−− MILLIMETERS b e e1 0.696 BSC 1.960 ÈÈÈ ÈÈÈ A B PIN 1 INDICATOR 0.05 C 0.13 C A1 C 0.076 0.180

0.914 BSC

*For additional information on our Pb−Free strategy and soldering details, please download the ON Semiconductor Soldering and Mounting Techniques Reference Manual, SOLDERRM/D. SOLDERING FOOTPRINT* TOP VIEW SIDE VIEWNOTE 3 RECOMMENDED AA2 E 3.683 BSC

1.067 BSC

−−− 1.780 0.478 0.538 L2 1.177 1.237 D D2 0.203 BSC

6.350 BSC

0.13 C 0.13 C D E b1 0.275 0.335 e2 0.699 BSC L 0.580 0.640 e A0.05 BC 0.03 C 25X b BOTTOM VIEW 12 20 e/2 NOTE 4 A0.05 BC 0.03 C 26X L NOTE 4 6X b1 6X b1 0.540 25X DIMENSIONS: MILLIMETERS 0.699 PACKAGE OUTLINE PITCH 1 0.370 0.203 0.914 0.370 1.067 1.239 0.696 PITCH 0.650 26X PITCH

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