R3910

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© Semiconductor Components Industries, LLC, 2015 August, 2015 − Rev. 9

1 Publication Order Number:

Description

RHYTHM/C0116 R3910 is a preconfigured hearing health processor based on a powerful DSP platform. Featuring iSceneDetect /C0116 environmental classification, adaptive noise reduction, superior feedback cancellation, fully automated and adaptive microphone directionality, and up to 8−channel WDRC, the R3910 is ideal for high−end, full featured products. Available in one of the industry’s smallest form−factors, it is well suited for all hearing aid types, including those placed deep in the ear canal. 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/C0116 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 /C0116 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 Canceller − Automatically reduces acoustic feedback. It allows for an increase in the stable gain while minimizing artifacts for music and tonal input signals. Adaptive Noise Reduction − The adaptive noise algorithm on R3910 monitors noise levels independently in 128 individual bands and employs advanced psychoac oustic models to provide user comfort. Tinnitus Masking − R3910 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 Tone 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. www.onsemi.com

25 PAD

R3910−CFAB XXXXXX R3910−CFAB = Specific Device Code XXXXXX = Work Order Number (Bottom View) 8910 17 18 VIN1 N/C N/C N/C N/C N/C 25 24 VIN2 TIN DAI VC D_VC SDA CLK MS1 VREG MGND GND PGND OUT+ OUT− VBP VB MS2 N/C N/C See detailed ordering and shipping information on page 18 of this data sheet.

ORDERING INFORMATION

www.onsemi.com Other Key Features − R3910 also supports the following features: FrontWave ® 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 enhancements, optional peak clipping, flexible compression adjustments, direct interfaces to analog or digital volume control, rocker switch, direct audio input and telecoil. R3910 also encompasses industry−leading security features to avoid cloning and software piracy.

Features

• Advanced Research Algorithms: ♦ iSceneDetect Environmental Classification ♦ Automatic Adaptive Directional Microphones (ADM) ♦ FrontWave Directionality ♦ 128−band Adaptive Noise Reduction ♦ Adaptive Feedback Cancellation (AFC)
• 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/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 Digital V olume Control with Programmable Range
• Rocker Switch Support
• Support for Active Hi or Active Lo Switching
• 20−bit Audio Processing
• thinSTAX Packaging
• E1 RoHS Compliant Hybrid
• These Devices are Pb−Free and are RoHS Compliant thinSTAX/C0041 Packaging
• Hybrid Typical Dimensions: 0.220 x 0.125 x 0.060 in. (5.59 x 3.18 x 1.52 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 (Supply Voltage VB = 1.25 V; Temperature = 25°C)

100 Hz − 8 kHz

Table 2. ELECTRICAL CHARACTERISTICS (Supply Voltage VB = 1.25 V; Temperature = 25°C) (continued) 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 RHYTHM R3910 OVERVIEW R3910 is a programmable multi−processor DSP platform implemented on a thin−stacked package. This DSP platform is the hearing industry’s first 90 nm Silicon−on−Chip platform enabling design of highly−efficient and flexible hearing aid solutions. The multi−processor DSP system maximizing MIPS/ /C0109W with a unique reconfigurable architecture, integrated high−resolution dual ADC and a single DAC available in miniaturized package sizes, offering unmatched DSP processing capability and flexibility in an ultra small footprint with best in the industry power consumption. R3910 incorporates industry leading hearing algorithms allowing for easy integration into a wide range of hearing products. The DSP core implements FrontWave directional processing, programmable filters, adaptive algorithms, compression, wideband gain, and volume control. The adaptive algorithms include Adaptive Noise Reduction, Adaptive Feedback Cancellation and Automatic Adaptive Directional Microphones. Adaptive Noise Reduction reduces audible noise in a low distortion manner while preserving perceived speech levels. The Adaptive Feedback Canceller reduces acoustic feedback while offering robust performance against pure tones. The Adaptive Directional Microphone algorithm automatically reduces the level of sound sources that originate from behind or from the side of the hearing−aid wearer without affecting sounds from the front. Additionally, the Automatic Adaptive Directional Microphones algorithm automatically reduces current by turning off the second input channel if it is not needed. The iLog 4.0 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 user and fine tune the fitting. iSceneDetect 1.0 is an classification algorithm that senses the users environment and automatically optimizes the hearing aid to maximize user comfort and audibility in that environment without any user interaction. R3910 supports iSceneDetect in 1 mic omni, static directional or adaptive directional modes. R3910 comes with Evoke advanced acoustic indicators. Evoke allows manufacturers to provide more complex, multi−frequency tones, in addition to traditional programmable tones for memory changes and low battery indication, which can simulate musical notes or chords. R3910 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. R3910 delivers advanced features and enhanced performance previously unavailable to a product in its class. As well, R3910 contains security features to protect clients’ intellectual property against device cloning and software piracy. SIGNAL PATH 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. 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. In iSceneDetect, FrontWave, ADM or Automatic ADM operation, a multi−microphone signal is used to produce a directional hearing aid 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 32 kHz 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. 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. clipper) as shown in Figure 1. 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. input. The forward path of the hearing aid is not affected. a time−domain model of the feedback path. completely automatic operation. Figure 5. Adaptive Feedback Canceller (AFC) estimate of the feedback−path impulse response. attenuation gain independently in each of the 128 bands.

www.onsemi.com 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 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 The Adaptive Directional Microphone (ADM) algorithm from ON Semiconductor 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 R3910 has a datalogging function that records information every 4 seconds 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

• 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 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, 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 R3910 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 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. If the noise is injected before the VC and the audio path is also enabled, the device can be set up to either have both the audio path and noise adjust via the VC or to have the noise only adjust via the VC. If the noise is injected after the VC, it is not affected by VC changes (see Table 4).

Table 4. NOISE INJECTION EFFECT ON VC narrow−band noise is centred on an audiometric frequency. stimuli are individually adjustable. words at the system sampling rate of 32 kHz. post−A/D attenuation depending on the input level.

• 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

• 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 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 the Interactive Data Sheet (IDS), enabling manual adjustment of the telecoil gain compensation. Automatic Telecoil R3910 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. R3910 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 aid) 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 R3910 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

information related to the associated biquad. stable again, it can be re−enabled. microphone/telecoil compensation, low−frequency EQ, etc. software or controlled externally via a physical interface. functionality, although only one can be enabled at a time. also be used to control memory selects. how to wire the digital volume control to R3910. Figure 7. Wiring for Digital Volume Control be configured and selected by the MS switches on R3910.

• MSSmode: this mode determines whether a connected switch is momentary or static.
• Donly: this parameter determines whether the MS2 switch is dedicated to the last memory position. There are four MS switch modes of operation as shown in Table 5 below.

Table 5. MS Switch Modes

settings tab as shown in Table 6 below. Table 6. MS Switch Logic Levels vs. IDS PullUpDown Settings valid memory, a button press causes memory A to be loaded. parameter to ‘Momentary’ and ‘Donly’ to ‘disabled’. parameter to ‘Momentary’ and ‘Donly’ to ‘enabled’. If Pull−up/Pull-down = Pull-down and MS2 = HIGH: D... If Pull−up/Pull-down = Pull-up and MS2 = LOW: D... Table 7. DYNAMIC EXAMPLE WITH FOUR VALID MEMORIES

This mode uses two static switches to change memories. invalid, the part defaults to memory A. parameter to ‘static’ and ‘Donly’ to ‘disabled’. Table 8. MEMORY SELECTED BY STATIC SWITCH This mode uses two static switches to change memories. memories (even if four valid memories are programmed). parameter to ‘static’ and ‘Donly’ to ‘enabled’. Table 9. MEMORY SELECTED BY STATIC SWITCH memory before reaching the final memory. a memory is invalid, the part defaults to memory A. intermittent transients exceed the threshold.

www.onsemi.com 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 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 R3910 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 the average supply voltage is above 0.95 V , an instantaneous supply voltage fluctuation below 0.95 V will trigger an immediate 3 dB gain reduction. After the 3 dB gain reduction has been applied, the advanced reset model holds off checking the instantaneous voltage level for a monitoring period of 30 second in order to allow the voltage level to stabilize. If after the stabilization time the instantaneous voltage drops a second time below 0.95 V during the next monitoring period, the gain will be reduced an additional 3 dB for a 6 dB total reduction and a 30 second stabilization time is activated. The advanced reset mode continues to monitor the instantaneous voltage levels over 30 second monitoring periods. If the instantaneous voltage remains above 1.1 V during that monitoring period, the gain will be restored to the original setting regardless of whether one or two gain reductions are applied. If two gain reductions are applied and the instantaneous voltage level remains above 1.0 V for a monitoring period, the gain will be restored to a 3 dB reduction. Should the average supply voltage drop below 0.95 V , the device will then reduce the gain by 1 dB every 10 seconds until either the average supply voltage rises above 0.95 V or a total of 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 R3910 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

www.onsemi.com 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 ARK User’s Guide. SDA and I2C Communication R3910 can be programmed using the SDA or I 2C 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 R3910 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, R3910 is programmed via the SDA pin using industry standard programming boxes. I 2C mode is a two wire interface which uses the SDA pin for bidirectional data and CLK as the interface clock input. I2C programming support is available on the HiPro (serial or USB versions) and ON Semiconductor’s DSP Programmer 3.0. Power Supply Considerations R3910 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. R3910 hybrids have a separate power supply and ground connections for the output stage. This enables hearing aid 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. Input Connection and Layout Considerations It is recommended to connect unused audio input pins directly to MGND to minimize the possibility of noise pickup. Inputs are internally AC coupled, so there is no additional leakage current when inputs are connected directly to ground. In order to further minimize noise at the inputs the following guidelines are recommended:
• MGND is used as reference ground plane for input signals. All input components should be grounded to MGND. This ground plane should be isolated from all other ground connections in the system.
• Keep the input traces as short as possible and avoid routing traces near high noise sources such as the OUT+ and OUT− pins
• Star ground input component grounds to the MGND connection.

Device Package Shipping† R3910−CFAB−E1B 25 Pad Hybrid Case 127DN

25 Units / Bubble Pack

R3910−CFAB−E1T 25 Pad Hybrid Case 127DN

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 Evaluation Board for R3910 contact your Sales Account Manager or FAE and use part number SA3400GEVB.

Table 10. PAD POSITION AND DIMENSIONS

www.onsemi.com PACKAGE DIMENSIONS ÈÈÈ ÈÈÈ SIP25, 5.59x3.18 CASE 127DN ISSUE O SEATING PLANE NOTES: 1. DIMENSIONING AND TOLERANCING PER ASME Y14.5M, 1994. 2. CONTROLLING DIMENSION: MILLIMETERS. 3. COPLANARITY APPLIES TO THE SPHERICAL CROWNS OF THE PADS. DIM A MIN MAX −−− MILLIMETERS A2 −−− 1.65 e1 0.051 BSC 1.83 E D A B PIN A1 INDICATOR C 0.08 0.18 TOP VIEW SIDE VIEW A E 5.59 BSC e 0.686 BSC 0.13 C 0.13 C 0.05 C 0.13 C NOTE 3

1.092 BSC

1.067 BSC

L 0.554 0.614 L2 0.808 0.868 b 0.478 0.538 b1 0.427 0.487 D 3.18 BSC L3 0.275 0.335 23X b B C BOTTOM VIEW 123 A e A0.05 C NOTE 4 B 0.03 C 567 8 e/2 e e/2 2X b1 DETAIL A L2e L20X *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* RECOMMENDED DIMENSIONS: MILLIMETERS 0.614 0.686 PITCH 1.067 DETAIL B 20X 0.487 1.092 0.051 0.538 23X DETAIL B 0.686 0.335 0.686 0.868

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