SA3286

DATASHEET SEARCH SITE | WWW.ALLDATASHEET.COM

Document overview

• Manufacturer or author: Provided By ALLDATASHEET.COM(FREE DATASHEET DOWNLOAD SITE)
• PDF pages: 16

Technical content

© Semiconductor Components Industries, LLC, 2015 July, 2015 − Rev. 2

1 Publication Order Number:

INSPIRIA SA3286 features iSceneDetect /C0116 environmental classification, 128 −bands of noise reducti on, superior feedback cancellation and built−in feedback path measurement capabilities. The iSceneDetect Environment Classification algorithm automatically senses the hearing aid wearer’s environment and dynamically adjusts hearing enhancement algorithms (such as Feedback cancellation, noise reduction, compression, etc.) without user involvement. Environment classification on Inspiria SA3286 enables programming of hearing aid to a single program setting which can be employed by the hearing−aid wearer in all environments instead of manually changing to a different memory with a change in the acoustic environment. The Inspiria SA3286 comes with EVOKE /C0116 advanced acoustic indicators. EVOKE allows manufacturers to provide more complex, multi−frequency tones which can simulate musical notes or chords. The Inspiria SA3286 iLog/C0116 2.0 Datalogging feature records various parameters every 4 seconds to 60 minutes (programmable) during use of the device. Once these parameter values are read from the device, they can be used to counsel the hearing aid wearer and fine tune the fitting. The Inspiria SA3286’s Adaptive Noise Reduction monitors noise levels independently in 128 individual bands and employs advanced psychoacoustic models to provide user comfort. Based on a phase cancellation method, Inspiria SA3286’s adaptive feedback reduction algorithm provides an increase in added stable gain. It features rapid adjustment for dynamic feedback situations and resistance to tonal inputs. Automatic Adaptive Directional Microphone (ADM) algorithm from ON Semiconductor automatically reduces the level of sound sources that originate from behind or the side of the hearing −aid wearer without affecting sounds from the front by adjusting the null in the microphone polar pattern to minimize the noise level at the output of the ADM. To reduce current consumption, the algorithm can switch automatically between 1 mic omni and ADM depending on the acoustic environment. The Inspiria SA3286 is equipped with a noise source that can be used in treating tinnitus. The Tinnitus Treatment noise can be shaped and attenuated and then summed into the audio path either before or after the volume control. The Narrow−band Noise Stimulus feature allows the user to generate stimuli from the device that can be used for in situ audiometry. In addition to these adaptive algorithms, the Inspiria SA3286 also supports the following features: up to 8 channel WDRC, FRONTW A VE ® directional processing, cross fading between audio paths for click −free memory changes, 16 −band graphic equalizer, 8 generic biquad filters (configurable as parametric or other filter types), programming speed enhancements, optional peak clipping, flexible compression adjustments, volume control, rocker switch, and industry−leading security features to avoid cloning and software piracy. www.onsemi.com

16 PAD

OUT−OUT+ MGND PAD CONNECTION FMIC RMIC VC SDA GND PGND 16 1 MARKING DIAGRAM SA3286−E1 XXXXXX SA3286 = Specific Device Code E1 = RoHS Compliant Hybrid XXXXXX = Work Order Number (Bottom View) See detailed ordering and shipping information on page 14 of this data sheet.

ORDERING INFORMATION

www.onsemi.com

Features

• Advanced Research ♦ 128−band Adaptive Noise Reduction ♦ Adaptive Feedback Cancellation ♦ Feedback Path Measurement Tool ♦ Automatic Adaptive Directional Microphones ♦ Environmental Classification
• iSceneDetect Environmental Classification 1.0
• iLog Datalogging 2.0
• Tinnitus Treatment
• EVOKE Acoustic Indicators
• Auto Telecoil with Programmable Delay
• FrontWave Directional Processing
• 1, 2, 4, 6 or 8 Channel WDRC Compression
• AGC−O with Variable Threshold, Time Constants, and Optional Adaptive Release
• 16−band Graphic EQ
• Narrow−band Noise Stimulus
• Optimized Programming Speed
• 8 Biquadratic Filters
• Four Analog Inputs
• 16 kHz or 8 kHz Bandwidth
• Four Fully Configurable Memories with Audible Memory Change Indicator
• 93 dB Input Dynamic Range with HRX/C0116 Headroom Extension
• 128−bit Fingerprint Security System and Other Security Features to Protect against Device Cloning and Software Piracy
• High Fidelity Audio CODEC
• Soft Acoustic Fade between Memory Changes
• Drives Zero−bias 2−terminal Receivers
• Internal or External Analog or Digital V olume Control with Programmable Range
• Rocker Switch Support
• 20−bit Audio Precision
• thinSTAX® Packaging
• E1 RoHS Compliant Hybrid thinSTAX Packaging
• Hybrid typical dimensions: 0.215 x 0.124 x 0.067 in. (5.46 x 3.15 x 1.70 mm)

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 (VBAT = 1.25 V; Temperature = 25°C) (Note 2)

100 Hz − 8 KHz

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

1. Total system gain consists of: wideband system gain + channel gain + converter gain. T otal System gain is calibrated during Cal/Config process.
2. Average currents with auto −ADM or iSceneDetect modes enabled are typically lower.

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

1. Total system gain consists of: wideband system gain + channel gain + converter gain. T otal System gain is calibrated during Cal/Config process.
2. Average currents with auto −ADM or iSceneDetect modes enabled are typically lower.

Figure 4. Typical Hearing Instrument Assembly Diagram hearing−aid wearer without affecting sounds from the front. turning off the second input channel if it is not needed. to counsel the user and fine tune the fitting. the audio path either before or after the volume control. device cloning and software piracy.

www.onsemi.com 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. 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 V olume Control (between the Wideband Gain and the V olume Control) or after the Volume Control (between post 4 and the Peak Clipper) as shown in Figure 1. Functional Block Description 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 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 seven system events: memory select (A, B, C or D), 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 Inspiria SA3286’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 5) allows for an increase in the stable gain1 of the hearing instrument while minimizing artefacts for music and tonal input signals. As with previous products, the feedback canceller provides completely automatic operation. (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.) 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) 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 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 ON Semiconductor’s Adaptive Directional Microphone (ADM) 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

www.onsemi.com 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
• V olume 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.

www.onsemi.com 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. For more information on filter design refer to the Biquad Filters In PARAGON Digital Hybrid information note. Rocker Switch The Inspiria SA3286 is equipped with a rocker switch feature that can perform both volume control (VC) adjustments or an audio memory switch. There are 3 modes of operation:

• 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 V olume Control (VC) can be either external or programmable. If external VC operation is selected, a further decision is required as to whether a variable resistor (analog VC) or a Digital V olume Control (DVC) will be connected to the 9−bit A/D converter. Analog Volume Control The external VC can be configured to work with either a two−terminal 200 k/C0087 variable resistor or a three −terminal 0.1 M/C0087 − 1M /C0087 variable resistor. In two −terminal configuration, the VC is connected between GND and the VC input. In three −terminal configuration, it is connected between GND, Vreg and the VC input. If using a two−terminal VC, it must be calibrated before use. Calibration is not necessary with a three −terminal connection. Hysteresis is built into the VC circuitry to prevent unintentional volume level toggling. A log taper potentiometer is recommended so that gain in dB would be linear with potentiometer rotation. The range of VC is adjustable and can be set between 1 dB (min) and 42 dB (max). Digital Volume Control If using a Digital V olume Control with the Inspiria SA3286, a resistor must be connected between the VC input and Vreg, and another resistor of the same value must be connected between the VC input and GND. The values of both resistors can be between 50 k/C0087 and 0.5 M/C0087. 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 6 dB and 42 dB in 6 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.

tremendous flexibility in switching between configurations. select options are selectable via the settings tab in IDS. memory, a button press causes memory A to be loaded. parameter to ‘Momentary’ and ‘Donly’ to ‘disabled’. parameter to ‘Momentary’ and ‘Donly’ to ‘enabled’. Table 3. DYNAMIC EXAMPLE WITH FOUR VALID MEMORIES (T = momentary switch is toggled; 0 = OPEN; 1 = HIGH) This mode uses two static switches to change memories. invalid, the part defaults to memory A. parameter to ‘static’ and ‘Donly’ to ‘disabled’. Table 4. MEMORY SELECTED IN STATIC SWITCH ON This mode uses two static switches to change memories. last valid memory when the static switch on MS2 is HIGH. memories (even if four valid memories are programmed). parameter to ‘static’ and ‘Donly’ to ‘enabled’. Table 5. MEMORY SELECTED IN STATIC SWITCH ON

www.onsemi.com In this mode, it is possible to jump from any memory to any other memory simply by changing the state of both switches. If both switches are changed simultaneously, then the transition is smooth. Otherwise, if one switch is changed and then the other, the part transitions to an intermediate memory before reaching the final memory. When MS2 is set HIGH, the state of the switch on MS is ignored. This prevents memory select beeps from occurring if switching MS when MS2 is HIGH. The part starts in whatever memory the switches are selecting. If a memory is invalid, the part defaults to memory A. AGC−O and Peak Clipper The output compression−limiting block (AGC−O) is an output limiting circuit whose compression ratio is fixed at ∝:1. The threshold level is programmable. The AGC −O module has programmable attack and release time constants. The AGC−O on the Inspiria SA3286 has optional adaptive release functionality. When this function is enabled, the release time varies depending on the environment. In general terms, the release time becomes faster in environments where the average level is well below the threshold and only brief intermittent transients exceed the threshold. Conversely, in environments where the average level is close to the AGC−O threshold, the release time applied to portions of the signal exceeding the threshold is longer. The result is an effective low distortion output limiter that clamps down very quickly on momentary transients but reacts more smoothly in loud environments to minimize compression pumping artifacts. The programmed release time is the longest release time applied, while the fastest release time is 16 times faster. For example, if a release time of 128 ms is selected, the fastest release time applied by the AGC −O block is 8 ms. The Inspiria SA3286 also includes the Peak Clipper block for added flexibility. Memory Switch Fader To minimize potential loud transients when switching between memories, the Inspiria SA3286 uses a memory switch fader block. When the memory is changed, the audio signal is faded out, followed by the memory select acoustic indicators (if enabled), and after switching to the next memory, the audio signal is faded back in. The memory switch fader is also used when turning the Tone Generator on or off, and during SDA programming. Power−On/Power−Off Behaviour and Low Battery Indicator During power−on, the Inspiria SA3286 hybrid is held in a reset state until the supply voltage (Vb) reaches a turn−ON threshold. A small portion of the hybrid’s internal control logic turns on and monitors the voltage to determine if the supply is stable. Once the supply is stable, the entire hybrid is activated and loads its configuration. Finally, the audio output turns on by smoothly transitioning to the expected output level. During normal operation, when a low battery condition is detected, the Inspiria SA3286 hybrid plays out a configurable acoustic indicator to indicate that the battery is low. This is repeated every five minutes until the device reaches the turn−OFF threshold. The low battery threshold is programmable in IDS between 1.0 V and 1.2 V in 10 mV increments. If V b drops below the turn −OFF threshold, then the Inspiria SA3286 hybrid is returned to its reset state and the audio output is muted. After a reset due to a low battery or a sudden supply transient, the recovery behaviour of the Inspiria SA3286 is determined by the selectable reset mode through ARKonline. There are four selectable reset modes as follows:

• Shallow−reset mode − After a low battery shutdown or transient shutdown, it allows the Inspiria SA3286 hybrid to immediately restart when the supply voltage rises above the turn−ON threshold. The device restarts in the memory that was last active when the shut down occurred. In summary, the device functions until the supply voltage drops below the turn−OFF threshold, and recovers when the device rises above the turn−ON threshold again.
• Deep−reset mode − After a low battery shutdown or transient shutdown, it does not allow the Inspiria SA3286 hybrid to restart. When a shutdown occurs (i.e., the supply voltage drops below the turn−OFF threshold), the device remains off until the supply voltage drops below approximately 0.3 V and subsequently rises above the turn−ON threshold. For the supply to drop below 0.3 V , the battery should be disconnected. Upon reconnecting the battery (preferably a new battery) the supply voltage rises above the turn−ON threshold, and depending if the supply is stable, the device restarts.
• Mixed mode − A combination of the first two modes. The device starts up in shallow−reset mode initially, then transitions to deep reset mode after five minutes.
• Advanced reset mode (recommended) − A more advanced combination of the first two modes, plus some additional intelligence. The device starts up in shallow−reset mode initially, so that after a low battery shutdown or a transient shutdown, the device immediately restarts when the supply voltage rises above the turn−ON threshold. When the device restarts, deep−reset mode is applied and the device operates in the memory that was last active when the shut down occurred. Additionally, the maximum output level is reduced through a 2 dB reduction of the AGCo and peak clipper. This operating condition is defined as transient reboot mode. The device operates in transient reboot mode (i.e., deep−reset mode and maximum

www.onsemi.com output reduction are applied) while monitoring the supply voltage. If the supply voltage remains above the turn−ON threshold for at least 30 secs, the device is allowed to exit transient reboot mode. The device returns to shallow−reset mode and the maximum output is restored. Generally, any low battery shutdown or transient shutdown that occurs while in shallow−reset mode (or while in the shallow −reset mode component of mixed mode or advanced reset mode) results in the Inspiria SA3286 hybrid restarting into the memory that was last active when the shut down occurred. The Inspiria SA3286 hybrid has this memory restart capability for up to three memories. A restart in any memory beyond the first three memories causes the device to restart in the initial memory, similar to the behaviour when a battery is first connected. The advanced reset mode described above also applies to up to three memories. Any additional memories would use the shallow−reset mode behaviour, and would restart in the initial memory after a shutdown. In any of the above reset modes, the Inspiria SA3286 hybrid can be configured through ARKonline to reduce the gain as the battery voltage drops. When the supply voltage falls below the low battery threshold, low battery tones are emitted and the wideband gain is reduced by 3 dB. As the battery voltage continues to drop, the low battery tones continue and the wideband gain continues to be reduced. Once the turn−OFF threshold is reached, the device shuts down. Software and Security The Inspiria SA3286 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. 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 Communication The Inspiria SA3286 is programmed via the SDA pin using industry standard programming boxes. 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 Inspiria SA3286 resulting in increased communication speed. Certain parameters in ARKonline can be selected to reduce the number of pages that need to be read out. Power Management The Inspiria SA3286 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 Inspiria SA3286 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.

Device Package Shipping† SA3286−E1 16 Pad Hybrid 25 Units / Bubble Pack SA3286−E1−T 16 Pad Hybrid 500 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 Evaluation Board for Inspiria SA3286 contact your Sales Account Manager or FAE and use part number GA3280GEVB.

Table 6. PAD POSITION AND DIMENSIONS

www.onsemi.com PACKAGE DIMENSIONS 0.026 (0.660) 0.016 (0.406) 0.215 (5.46)

0.072 MAX

(1.83) 0.124 (3.15) Dimension units are in inches. Dimensions in parentheses are in millimeters, converted from inches and include minor rounding errors. 1.000 inches = 25.4mm Dimension tolerances: ±0.005 (±0.13) unless otherwise stated.

• = location of Pin 1 RoHS compliant hybrid, MSL#4, 240°C peak reflow, SAC305. This Hybrid is designed for either point−to−point manual soldering or for reflow according to ON Semiconductor’s reflow process. SA3286 − E1 XXXXXX ON Semiconductor and the are registered trademarks of Semiconductor Components Industries, LLC (SCILLC) or its subsidia ries in the United States and/or other countries. SCILLC owns the rights to a number of patents, trademarks, copyrights, trade secrets, and other intellectual property. A listing of SCILLC’s product/patent coverage may be accessed or guarantee regarding the suitability of its products for any particular purpose, nor does SCILLC assume any liability arising out of the application or use of any product or circuit, and specifically disclaims any and all liability, including without limitation special, consequential or incidental damages. “Typical” parameters which may be provided in SCILLC data sheets and/or specifications can and do vary in different applications and actual performance may vary over time. All operating parameters, including “Typicals” must be validated for each customer application by customer’s technical experts. SCILLC does not convey any license under its patent rights nor the rights of others. SCILLC products are not designed, intended, or authorized for use as components in systems intended for surgical implant into the body, or other applications intended to support or sustain life, or for any other application in which the failure of the SCILLC product could create a situation where personal injury or death may occur. Should Buyer purchase or use SCILLC products for any such unintended or unauthorized application, Buyer shall indemnify and hold SCILLC and its officers, employees, subsidiaries, affiliates, and distributors harmless against all claims, costs, damages, and expenses, and reasonable attorney fees arising out of, directly or indirectly, any claim of personal injury or death associated with such unintended or unauthorized use, even if such claim alleges that SCILLC was negligent regarding the design or manufacture of the part. SCILLC is an Equal Opportunity/Affirmative Action Employer. This literature is subject to all applicable copyright laws and is not for resale in any manner. PUBLICATION ORDERING INFORMATION N. American Technical Support: 800−282−9855 Toll Free USA/Canada Europe, Middle East and Africa Technical Support: Phone: 421 33 790 2910 Japan Customer Focus Center Phone: 81−3−5817−1050 SA3286/D iSceneDetect, iLog, EVOKE and HRX are trademarks of Semiconductor Components Industries, LLC. thinSTAX, FRONTWAVE and ARKonline are registered trademarks of Semiconductor Components Industries, LLC. LITERATURE FULFILLMENT: Literature Distribution Center for ON Semiconductor P.O. Box 5163, Denver, Colorado 80217 USA Phone: 303−675−2175 or 800−344−3860 Toll Free USA/Canada Fax: 303−675−2176 or 800−344−3867 Toll Free USA/Canada Email: orderlit@onsemi.com ON Semiconductor Website: www.onsemi.com Order Literature: http://www.onsemi.com/orderlit For additional information, please contact your local Sales Representative