Rhythm R3710预配置DSP系统实践应用指南

RHYTHM R3710信号路径

There are two inputs into the audio signal path. The first input is the front microphone and the second input can be a second microphone or telecoil input as selected by a programmable MUX. The front microphone input is intended as the main microphone audio input.

 

 

Figure 3. Test Circuit

 

 

Figure 4. Typical Application Circuit

The audio input is buffered, sampled and converted into digital form using an A/D converter. The digital output is converted into a selectable 32 kHz or 16 kHz, 20−bit digital audio signal. Further IIR filter blocks process the microphone signal. These are 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 Volume Control (between the Wideband Gain and the Volume Control) or after the Volume Control (between post 4 and the Peak Clipper) as shown in Figure 1.

功能模块描述

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, music, quiet and noise environments and make the necessary adjustments to the parameters in the audio path, such as ANR, WDRC and FBC, in order to optimize the hearing aid settings for thespecific environment.

iSceneDetect will gradually make the adjustments so the change in settings based on the environment is smooth and virtually unnoticeable. This feature will enable the hearing aid wearer to have an instrument which will work in any environment with a single memory.

Evoke Advanced Acoustic Indicators

Advanced acoustic indicators provide alerting sounds that are more complex, more pleasing and potentially more meaningful to the end user than the simple tones used on previous products. The feature is capable of providing pulsed, multi−frequency pure tones with smooth on and off transitions and also damped, multi−frequency tones that can simulate musical notes or chords.

A unique indicator sound can be assigned to each of the ten system events: memory select (A, B, C, 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, Rhythm R3710’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 gain (see Note) of the hearing instrument while minimizing artefacts for music and tonal input signals. As with previous products, the feedback canceller provides completely automatic operation.

NOTE: Added stable gain will vary based on hearing aid

style and acoustic setup. Please refer to the

Adaptive Feedback Cancellation Information

note for more details.

 

 

Figure 5. Adaptive Feedback Canceller (AFC)

Block Diagram

Feedback Path Measurement Tool

The Feedback Path Measurement Tool uses the onboard feedback cancellation algorithm and noise generator to measure the acoustic feedback path of the device. The noise generator is used to create an acoustic output signal from the hearing aid, some of which leaks back to the microphone via the feedback path. The feedback canceller algorithm automatically calculates the feedback path impulse response by analyzing the input and output signals. Following a suitable adaptation period, the feedback canceller coefficients can be read out of the device and used as an estimate of the feedback−path impulse response.

Adaptive Noise Reduction

The noise reduction algorithm is built upon a high resolution 128−band filter bank enabling precise removal of noise. The algorithm monitors the signal and noise activities in these bands, and imposes a carefully calculated attenuation gain independently in each of the 128 bands. The noise reduction gain applied to a given band is determined by a combination of three factors:

Signal−to−Noise Ratio (SNR)

Masking threshold

Dynamics of the SNR per band

The SNR in each band determines the maximum amount of attenuation to be applied to the band . the poorer the SNR, the greater the amount of attenuation. Simultaneously, in 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.

In−Situ Datalogging − iLog 6.0

Rhythm R3710 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

Volume 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:[!--empirenews.page--]

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, the 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

Rhythm R3710 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 attention 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. The noise can also be duty cycled. The on and off time of the noise stimulus can be adjusted so that the on time is from 1 − 30s as well as the off time. An off time set to 0s turns off the duty cycling.

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 in injected after the VC, it is not affected by VC changes (see Table 4).

 

 

Table 4. NOISE INJECTION EFFECT ON VC

Narrow−band Tone and Noise Stimulus

Rhythm R3710 is capable of producing Narrow−band Noise and Tone 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 A/D converter is a second order sigma−delta modulator operating at a 2.048 MHz sample rate. The system’s audio input is pre−conditioned with antialias filtering and a programmable gain pre−amplifier. This analog output is over−sampled and modulated to produce a 1−bit Pulse Density Modulated (PDM) data stream. The digital PDM data is then decimated down to Pulse−Code Modulated (PCM) digital words at the system sampling rate of 32 kHz.

The D/A is comprised of a digital, third order sigma−delta modulator and an H−bridge. The modulator accepts PCM audio data from the DSP path and converts it into a 64−times or 128−times over−sampled, 1−bit PDM data stream, which is then supplied to the H−bridge. The H−bridge is a specialized CMOS output driver used to convert the 1−bit data stream into a low−impedance, differential output voltage waveform suitable for driving zero−biased hearing aid receivers.

HRX Head Room Expander

Rhythm R3710 has an enhanced Head Room Extension (HRX) circuit that increases the input dynamic range of Rhythm R3710 without any audible artifacts. This is accomplished by dynamically adjusting the pre−amplifier’s gain and the post−A/D attenuation depending on the input Level.

Channel Processing

Figure 6 represents the I/O characteristic of independent

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

the following main regions:

Low input level expansion (squelch) region

Low input level linear region

Compression region

High input level linear region (return to linear)

 

 

Figure 6. Independent Channel I/O Curve Flexibility

The I/O characteristic of the channel processing can be

adjusted in the following ways:

Squelch threshold (SQUELCHTH)

Low level gain (LLGAIN)

Lower threshold (LTH)

High level gain (HLGAIN)

Upper threshold (UTH)

Compression ratio (CR)

To ensure that the I/O characteristics are continuous, it is necessary to limit adjustment to a maximum of four of the last five parameters. During Parameter Map creation, it is 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.

Graphic Equalizer

Rhythm R3710 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:

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:

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.

Volume Control and Switches

External Volume Control

The volume of the device can either be set statically via software or controlled externally via a physical interface.

Rhythm R3710 supports both analog and digital volume control functionality, although only one can be enabled at a time. Digital control is supported with either a momentary switch or a rocker switch. In the latter case, the rocker switch can also be used to control memory selects.

Analog Volume Control

The external volume control works with a three−terminal 100 k

– 360 k

variable resistor. The volume control can have either a log or linear taper, which is selectable via software. It is possible to use a VC with up to 1 M

of resistance, but this could result in a slight decrease in the resolution of the taper.

Digital Volume Control

The digital volume control makes use of two pins for volume control adjustment, VC and D_VC, with momentary switches connected to each. Closure of the switch to the VC pin indicates a gain increase while closure to the D_VC pin indicates a gain decrease. Figure 7 shows how to wire the digital volume control to Rhythm R3710.

The digital volume control can be setup to adjust both volume levels and memory configurations depending on the length of time the momentary switch is depressed.

It is also possible to read and write the digital volume control with the ARK software. Using these software functions will lock out the digital volume control until the next time the hybrid is powered on.

 

 

Figure 7. Wiring for Digital Volume Control

Memory Select Switches

One or two, two−pole Memory Select (MS) switches can be used with Rhythm R3710. This enables users tremendous flexibility in switching between configurations. These switches may be either momentary or static and are configurable to be either pull−up or pull−down through the settings tab in IDS.

Up to four memories can be configured on Rhythm R3710. Memory A must always be valid. All memory select options are selectable via the settings tab in IDS.[!--empirenews.page--]

Momentary Switch on MS1

This mode uses a single momentary switch on MS1 (Pin 10) to change memories. Using this mode causes the part to start in memory A, and whenever the button is pressed, the next valid memory is loaded. When the user is in the last valid memory, a button press causes memory A to be loaded.

This mode is set by programming the ‘MSSMode’ parameter to ‘Momentary’ and ‘Donly’ to ‘disabled’.

Example:

If 4 valid memories: ABCDABCDA

If 3 valid memories: ABCABCA

If 2 valid memories: ABABA

If 1 valid memory: AAA

Momentary Switch on MS1, Static Switch on MS2 (Jump to Last Memory)

This mode uses a static switch on MS2 (Pin 9) and a momentary switch on MS1 (Pin 10) to change memories. If the static switch is OPEN, the part starts in memory A and the momentary switch is enabled, with the exception that memory D is not used. If the pull−up/pull−down resistors are set to pull−down, and the static switch on MS2 is set to HIGH, the part automatically jumps to memory D (occurs on startup or during normal operation). If the pull−up/pull−down resistors are set to pull−up, and the static switch on MS2 is set to LOW, the part automatically jumps to memory D (occurs on startup or during normal operation).

In the above setup when the static switch is CLOSED, the momentary switch is disabled, preventing memory select beeps from occurring. When MS2 is set to OPEN, the part returns to the last select memory.

This mode is set by programming the ‘MSSMode’ parameter to ‘Momentary’ and ‘Donly’ to ‘enabled’.

Example:

When MS2 = OPEN, then MS1 can cycle through up to 3 valid memories: ABCABCA…

If Pull−up/Pull−down = Pull−down and MS2 = HIGH: D, then memory D is enabled

If Pull−up/Pull−down = Pull−up and MS2 = LOW: D, then memory D is enabled

Table 5. DYNAMIC EXAMPLE WITH FOUR VALID MEMORIES AND MS2 PULL−UP/PULL−DOWN = PULL−DOWN (T = MOMENTARY SWITCH IS TOGGLED; 0 = OPEN; 1 = HIGH)

Static Switch on MS1 and MS2

This mode uses two static switches to change memories. Table 6 describes which memory is selected depending on the state of the switches.

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. The part starts in whatever memory the switches are selecting. If a memory is invalid, the part defaults to memory A.

This mode is set by programming the ‘MSSMode’ parameter to ‘static’ and ‘Donly’ to ‘disabled’.

Table 6. MEMORY SELECTED BY STATIC SWITCH ON MS1 AND MS2 MODE; INTERNAL RESISTORS SET TO PULL DOWN (EXAMPLE WITH FOUR VALID MEMORIES)

Static Switch on MS1, Static Switch on MS2 (Jump to Last Memory)

This mode uses two static switches to change memories. Unlike in the previous example, this mode will switch to the last valid memory when the static switch on MS2 is HIGH or LOW depending on the configuratoin of MS2. This means that this mode will only use a maximum of three memories (even if four valid memories are programmed). Tables 7 and 8 describe which memory is selected depending on the state of the switches.

This mode is set by programming the ‘MSSMode’ parameter to ‘static’ and ‘Donly’ to ‘enabled’.

Table 7. MEMORY SELECTED BY STATIC SWITCH ON MS1, STATIC SWITCH ON MS2 (JUMP TO LAST MEMORY) MODE; INTERNAL RESISTORS SET TO PULL−DOWN Table 8. MEMORY SELECTED BY STATIC SWITCH ON MS1, STATIC SWITCH ON MS2 (JUMP TO LAST MEMORY) MODE; INTERNAL RESISTORS SET TO PULL−UP 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.

With pull−up/pull−down = pull−down, when MS2 is set HIGH, the state of the switch on MS1 is ignored. This prevents memory select beeps from occurring if switching MS1 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. With pull−up/pull−down = pull−up, when MS2 is set LOW, the state of the switch on MS1 is ignored. This prevents memory select beeps from occurring if switching MS1 when MS2 is LOW. 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 Rhythm R3710 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.

Rhythm R3710 also includes the Peak Clipper block for added flexibility.

Memory Switch Fader

To minimize potential loud transients when switching between memories, Rhythm R3710 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 Management

Rhythm R3710 has three user−selectable power management schemes to ensure the hearing aid turns off gracefully at the end of battery life. Shallow reset, Deep reset and Advanced Reset mode. It also contains a programmable power on reset delay function.

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 Rhythm R3710 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

Rhythm R3710 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 and I2C Communication

Rhythm R3710 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 Rhythm R3710 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, Rhythm R3710 is programmed via the SDA pin using industry standard programming boxes. I2C mode is a two wire interface which uses the SDA pin for bidirectional data and CLK as the interface clock input. programming support is available on the HiPro (serial or USB versions) and ON Semiconductor’s DSP Programmer 3.0.

Power Supply Considerations

Rhythm R3710 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. Rhythm R3710 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.

 

 

Table 9. PAD POSITION AND DIMENSIONS (mil)