R3910 D

RHYTHM R3910
Preconfigured DSP System
for Hearing Aids
Description
© Semiconductor Components Industries, LLC, 2015
August, 2015 − Rev. 9
1
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25 PAD
HYBRID
CASE 127DN
PAD CONNECTION
17
18
VIN1
1
VREG
TIN
16
19
N/C
2
MGND
DAI
15
20
N/C
3
GND
VC
14
21
N/C
4
PGND
D_VC
13
22
N/C
5
OUT+
SDA
12
23
N/C
6
OUT−
CLK
11
7
VBP
MS1
10
25
24
N/C
VIN2
N/C
RHYTHMt R3910 is a preconfigured hearing health processor
based on a powerful DSP platform. Featuring iSceneDetectt
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.
iLogt 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.
E VO K E t A d v a n c e d A c o u s t i c I n d i c a t o r s − A l l o w s
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 psychoacoustic 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.
9
8
VB
MS2
(Bottom View)
MARKING DIAGRAM
R3910−CFAB
XXXXXX
R3910−CFAB = Specific Device Code
XXXXXX
= Work Order Number
ORDERING INFORMATION
See detailed ordering and shipping information on page 18 of
this data sheet.
Publication Order Number:
R3910/D
RHYTHM R3910
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 HRXt 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 Volume 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) Packaging
• Hybrid Typical Dimensions:
0.220 x 0.125 x 0.060 in.
(5.59 x 3.18 x 1.52 mm)
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2
RHYTHM R3910
BLOCK DIAGRAM
VB
8
ACOUSTIC
INDICATORS
VREG
1
VIN1
18
VIN2
17
TIN
16
DIA
15
POR
CIRCUITRY
VOLTAGE
REGULATOR
POST
BIQUAD
FILTERS
3&4
A/D
M
U
X
A/D
ADAPTIVE
DIRECTIONAL
MICROPHONE
or
FRONTWAVE
MIC/TCOIL
COMP
*
PEAK
CLIPPER
D/A
HBRIDGE
NOISE GENERATOR
AND SHAPER
FEEDBACK
CANCELLER
11
VOLUME
CONTROL
Σ
WIDEBAND
GAIN
CONTROL
A/D
6
OUT−
ENVIRONMENTAL
CLASSIFICATION
128 bands
FREQUENCY
BA N D
SYNTHESIS
PROGRAMMING
INTERFACE
Noise Reduction (128 bands)
DATA
LOGGING
Graphic EQ (16 bands)
CLOCK
GENERATOR
20
21
22
23
3
GND
* If Input Mode = 1 mic omni, mic + telecoil, mic + DAI
** If Input Mode = 2 mic omni, rear only, directional
Figure 1. Hybrid Block Diagram
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3
14
VC
13
DVC
10
MS1
9
MS2
POST
BIQUAD
FILTERS
1&2
CONTROL
(MS/DIGVC)
WDRC (1,2,4,6 or 8 channels)
19
OUT+
PGND
AGCO
FREQUENCY
BAND
ANALYSIS
SDA
CLK
5
4
2
12
VBP
**
PRE
BI Q UA D
FILTERS
MGND
CROSS
FADER
Σ
7
EEPROM
RHYTHM R3910
SPECIFICATIONS
Table 1. ABSOLUTE MAXIMUM RATINGS
Parameter
Value
Units
0 to +40
°C
−20 to +70
°C
50
mW
Maximum Operating Supply Voltage
1.65
VDC
Absolute Maximum Supply Voltage
1.8
VDC
Operating Temperature Range
Storage Temperature Range
Absolute Maximum Power Dissipation
Stresses exceeding those listed in the Maximum Ratings table may damage the device. If any of these limits are exceeded, device functionality
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)
Parameter
Symbol
Conditions
Min
Typ
Max
Units
Minimum Operating Supply Voltage
VBOFF
Ramp down, audio path
0.93
0.95
0.97
V
Ramp down, control logic
0.77
0.80
0.83
Ramp up, zinc−air
1.06
1.10
1.16
Supply Voltage Turn On Threshold
VBON
Hybrid Current
Ramp up, NiMH
1.16
1.20
1.24
All functions, 32 kHz sampling rate
−
665
−
All functions, 16 kHz sampling rate
−
575
−
V
mA
EEPROM Burn Cycles
−
−
100 k
−
−
cycles
Low Frequency System Limit
−
−
−
125
−
Hz
High Frequency System Limit
−
−
−
16
−
kHz
Total Harmonic Distortion
THD
VIN = −40 dBV
−
−
1
%
THDM
VIN = −15 dBV, HRX − ON
−
−
3
%
fCLK
−
3.973
4.096
4.218
MHz
VREG
−
0.87
0.90
0.93
V
PSRRSYS
1 kHz, Input referred, HRX enabled
−
70
−
dB
Input Referred Noise
IRN
Bandwidth 100 Hz − 8 kHz
−
−108
−106
dBV
Input Impedance
ZIN
1 kHz
−
3
−
MW
Anti−aliasing Filter Rejection
−
f = [DC − 112 kHz], VIN = −40 dBV
−
80
−
dB
Crosstalk
−
Between both A/D and Mux
−
60
−
dB
Maximum Input Level
−
−
−
−15
−13
dBV
mV
THD at Maximum Input
Clock Frequency
REGULATOR
Regulator Voltage
System PSRR
INPUT
Analogue Input Voltage Range
VAN_IN
VIN1, VIN2, Al
0
−
800
VAN_TIN
TIN
−100
−
800
Input Dynamic Range
−
HRX − ON Bandwidth
100 Hz − 8 kHz
−
95
96
dB
Audio Sampling Rate
−
−
8
−
48
kHz
−
100 Hz − 8 kHz
−
88
−
dB
ZOUT
−
−
10
13
W
OUTPUT
D/A Dynamic Range
Output Impedance
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RHYTHM R3910
Table 2. ELECTRICAL CHARACTERISTICS (Supply Voltage VB = 1.25 V; Temperature = 25°C) (continued)
Parameter
Symbol
Conditions
Min
Typ
Max
Units
Resolution (monotonic)
−
−
7
−
−
bits
Zero Scale Level
−
−
−
0
−
V
Full Scale Level
−
−
−
VREG
−
V
RVC
Three−terminal connection
200
−
1000
kW
−
−
−
−
42
dB
Logic 0 Voltage
−
−
0
−
0.3
V
Logic 1 Voltage
−
−
1
−
1.25
V
Stand−by Pull Up Current
−
Creftrim = 6
3
5
6.5
mA
Sync Pull Up Current
−
Creftrim = 6
748
880
1020
mA
Max Sync Pull Up Current
−
Creftrim = 15
−
1380
−
mA
Min Sync Pull Up Current
−
Creftrim = 0
−
550
−
mA
Logic 0 Current (Pull Down)
−
Creftrim = 6
374
440
506
mA
Logic 1 Current (Pull Up)
−
Creftrim = 6
374
440
506
mA
TSYNC
Baud = 0
237
250
263
ms
Baud = 1
118
125
132
Baud = 2
59
62.5
66
Baud = 3
29.76
31.25
32.81
Baud = 4
14.88
15.63
16.41
Baud = 5
7.44
7.81
8.20
Baud = 6
3.72
3.91
4.10
Baud = 7
1.86
1.95
2.05
CONTROL A/D
VOLUME CONTROL
Volume Control Resistance
Volume Control Range
PC_SDA INPUT
PC_SDA OUTPUT
Synchronization Time
(Synchronization Pulse Width)
Product parametric performance is indicated in the Electrical Characteristics for the listed test conditions, unless otherwise noted. Product
performance may not be indicated by the Electrical Characteristics if operated under different conditions.
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RHYTHM R3910
Table 3. I2C TIMING
Standard Mode
Fast Mode
Symbol
Min
Max
Min
Max
Units
Clock Frequency
fPC_CLK
0
100
0
400
kHz
Hold time (repeated) START condition. After this
period, the first clock pulse is generated.
tHD;STA
4.0
−
0.6
−
msec
LOW Period of the PC_CLK Clock
tLOW
4.7
−
−
−
msec
HIGH Period of the PC_CLK Clock
tHIGH
4.0
−
−
−
msec
Set−up time for a repeated START condition
tSU;STA
4.7
−
−
−
msec
Data Hold Time:
for CBUS Compatible Masters
for I2C−bus Devices
tHD;DAT
5.0
0 (Note 1)
−
3.45 (Note 2)
−
0 (Note 1)
−
0.9 (Note 2)
Data set−up time
tSU;DAT
250
−
100
−
nsec
Rise time of both PC_SDA and PC_CLK signals
tr
−
1000
20 + 0.1 Cb
(Note 4)
300
nsec
Fall time of both PC_SDA and PC_CLK signals
tf
−
300
20 + 0.1 Cb
(Note 4)
300
nsec
tSU;STO
4.0
−
0.6
−
nsec
tBUF
4.7
−
1.3
−
msec
Output fall time from VIHmin to VILmax with a bus
capacitance from 10 pF to 400 pF
tof
−
250
20 + 0.1 Cb
250
nsec
Pulse width of spikes which must be suppressed
by the input filter
tSP
n/a
n/a
0
50
nsec
Capacitive load for each bus line
Cb
−
400
−
400
pF
Parameter
Set−up time for STOP condition
Bus free time between a STOP and
START condition
msec
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 tHD;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 I2C−bus device can be used in a Standard−mode I2C−bus system, but the requirement tSU;DAT P250ns must then be met.
This will automatically be the case if the device does not stretch the LOW period of the PC_CLK signal. If such a device does stretch the
LOW period of the PC_CLK signal, it must output the next data bit to the PC_SDA line tr max + tSU;DAT = 1000 + 250 = 1250 ns (according
to the Standard−mode I2C−bus specification) before the PC_CLK line is released.
4. Cb = total capacitance of one bus line in pF.
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RHYTHM R3910
Figure 2. I2C Mode Timing
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RHYTHM R3910
TYPICAL APPLICATIONS
VB
8
ACOUSTIC
INDICATORS
POR
CIRCUITRY
VOLTAGE
REGULATOR
1
OUT
3k75
17
3k75
M
U
X
16
3k75
A/D
15
5
POST
BIQUAD
FILTERS
3&4
A/D
18
7
MIC/TCOIL
COMP
*
ADAPTIVE
DIRECTIONAL
MICROPHONE
or
FRONTWAVE
CROSS
FADER
Σ
PEAK
CLIPPER
D/A
HBRIDGE
NOISE GENERATOR
AND SHAPER
FEEDBACK
CANCELLER
LP FILTER
6
4
**
VOLUME
CONTROL
AGCO
Σ
WIDEBAND
GAIN
3k75
PRE
BI Q UA D
FILTERS
CONTROL
A/D
14
200k
POST
BIQUAD
FILTERS
1&2
ENVIRONMENTAL
CLASSIFICATION
13
2
128 bands
FREQUENCY
BAND
ANALYSIS
12
FREQUENCY
BA N D
SYNTHESIS
10
9
WDRC (1,2,4,6 or 8 channels)
PROGRAMMING
INTERFACE
11
CONTROL
(MS/DIGVC)
Noise Reduction (128 bands)
DATA
LOGGING
Graphic EQ (16 bands)
EEPROM
CLOCK
GENERATOR
19
20
21
22
23
3
* If Input Mode = 1 mic omni, mic + telecoil, mic + DAI
** If Input Mode = 2 mic omni, rear only, directional
Note: All resistors in ohms and all capacitors in farads, unless otherwise stated.
Figure 3. Test Circuit
VB
8
ACOUSTIC
INDICATORS
POST
BIQUAD
FILTERS
3&4
A/D
18
ADAPTIVE
DIRECTIONAL
MICROPHONE
or
FRONTWAVE
17
16
15
POR
CIRCUITRY
VOLTAGE
REGULATOR
1
M
U
X
A/D
MIC/TCOIL
COMP
*
CROSS
FADER
Σ
PEAK
CLIPPER
D/A
HBRIDGE
Zero Biased Receiver
4
**
AGCO
PRE
BI Q UA D
FILTERS
VOLUME
CONTROL
Σ
WIDEBAND
GAIN
CONTROL
A/D
ENVIRONMENTAL
CLASSIFICATION
13
128 bands
FREQUENCY
BAND
ANALYSIS
11
14
POST
BIQUAD
FILTERS
1&2
2
12
5
6
NOISE GENERATOR
AND SHAPER
FEEDBACK
CANCELLER
7
FREQUENCY
BA N D
SYNTHESIS
CONTROL
(MS/DIGVC)
9
WDRC (1,2,4,6 or 8 channels)
PROGRAMMING
INTERFACE
Noise Reduction(128 bands)
DATA
LOGGING
Graphic EQ (16 bands)
CLOCK
GENERATOR
19
20
21
22
23
3
* If Input Mode = 1 mic omni, mic + telecoil, mic + DAI
** If Input Mode = 2 mic omni, rear only, directional
Note: All resistors in ohms and all capacitors in farads, unless otherwise stated.
Figure 4. Typical Application Circuit
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8
10
EEPROM
RHYTHM R3910
RHYTHM R3910 OVERVIEW
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.
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/mW 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
SIGNAL PATH
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
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
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
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RHYTHM R3910
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.
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.
Adaptive Feedback Canceller
The Adaptive Feedback Canceller 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 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 aid 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.
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 aid which will work in any
environment with a single memory.
Feedback path
H
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, D, E or F), 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.
+
−
Σ
G
H’
Estimated feedback
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.
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RHYTHM R3910
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.
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.
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.
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).
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.
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.
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
• Volume control setting
• Program memory selection
• Environment
• Ambient sound level
• Length of time the hearing aid was powered on
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RHYTHM R3910
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.
Table 4. NOISE INJECTION EFFECT ON VC
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.
Noise Insertion
Modes
VC Controls
Noise Injected
Off
Audio
Off
Pre VC
Audio + Noise
Pre VC
Post VC
Audio
Post VC
Noise only Pre VC
Noise
Pre VC
Noise only Post VC
−
Post VC
Pre VC with Noise
Noise
Pre VC
Narrow−band Noise Stimulus
R3910 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.
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).
HRX Head Room Expander
R3910 has an enhanced Head Room Extension (HRX)
circuit that increases the input dynamic range of the R3910
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)
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RHYTHM R3910
0
OUTPUT LEVEL (dBV)
−20
−30 Low Level
−40 Gain
−50
Automatic Telecoil
High Level
Gain
−10
Compression
Ratio
Lower
Threshold
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.
Upper
Threshold
−60
−70
−80
Squelch
Threshold
−90
−100
−120 −110 −100 −90 −80 −70 −60 −50 −40 −30 −20
INPUT LEVEL (dBV)
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)
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.
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
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:
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.
H(z) + b0 ) b1
1 ) a1
z −1 ) b2
z −1 ) a2
z −2
z −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 ǒx 2 14Ǔ
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RHYTHM R3910
Digital Volume Control
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.
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 R3910.
GND
VC
D_VC
Figure 7. Wiring for Digital Volume Control
Memory Select Switches
One or two, two−pole Memory Select (MS) switches can
be used with R3910. This gives users tremendous flexibility
in switching between configurations. Up to six memories can
be configured and selected by the MS switches on R3910.
Memory A must always be valid. The MS switches are either
momentary or static and are fully configurable through IDS
in the IDS setting tab.
The MS switch behavior is controlled by two main
parameters in IDS:
• 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.
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.
R3910 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.
There are four MS switch modes of operation as shown in
Table 5 below.
Analog Volume Control
The external volume control works with a three−terminal
100 kW − 360 kW variable resistor. The volume control can
have either a log or linear taper, which is selectable via
software. It is possible to use a VC with up to 1 MW of
resistance, but this could result in a slight decrease in the
resolution of the taper.
Table 5. MS Switch Modes
MS Switch Mode
MS1 Switch
MS2 Switch
Max # of Valid Memories
Donly
MSSMode
Use
Mode 1
Momentary
None
6
Off
Momentary
Simplest configuration
Mode 2
Momentary
Static
6
On
Momentary
Jump to last memory
Mode 3
Static
Static
4
Off
Static
Binary selection of memory
Mode 4
Static
Static
3
On
Static
Jump to last memory
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RHYTHM R3910
The flexibility of the MS switches is further increased by
allowing the MS switches to be wired to GND or VBAT,
corresponding to an active low or active high logic level on
the MS pins. This option is configured with the
MSPullUpDown/MS2PullUpDown setting in the IDS
settings tab as shown in Table 6 below.
Table 6. MS Switch Logic Levels vs. IDS PullUpDown Settings
“PullUpDown” Setting in IDS
MS switch state
MS input logic level
Switch connection
Pulldown
CLOSED
HI
To VBAT
Pulldown
OPEN
LOW
To VBAT
Pullup
CLOSED
LOW
To GND
Pullup
OPEN
HI
To GND
Mode 2: Momentary Switch on MS1, Static Switch on
MS2 (Jump to Last Memory)
In the following mode descriptions, it is assumed that the
PullUpDown setting has been properly configured for the
MS switch wiring so that a CLOSED switch state is at the
correct input logic level.
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
behaves like momentary, with the exception that the highest
valid memory (F if 6 memories selected) is not used. If the
static switch on MS2 is set to CLOSED, the part
automatically jumps to the highest valid memory location
(occurs on startup or during normal operation). In this setup,
the momentary switch’s state is ignored, preventing memory
select beeps from occurring. When MS2 is set to OPEN, the
part loads in the memory location selected before MS2 was
closed.
This mode is set by programming the ‘MSSMode’
parameter to ‘Momentary’ and ‘Donly’ to ‘enabled’.
Mode 1: Momentary Switch on MS1
This mode uses a single momentary switch on MS1 (Pin
10) to change memories. When using this mode the part
starts 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’.
Mode 1 Example:
If 6 valid memories: ABCDEFABCDEF…
If 5 valid memories: ABCDEABCDE…
If 4 valid memories: ABCDABCDA…
If 3 valid memories: ABCABCA…
If 2 valid memories: ABABA…
If 1 valid memory: AAA…
Mode 2 Example:
If MS2 = OPEN and there are 6 valid memories: ABCEFABCEF…
If MS2 = OPEN and there are 5 valid memories: ABCEABCE…
If MS2 = OPEN and there are 4 valid memories: ABCABCA…
If MS2 = OPEN and there are 3 valid memories: ABABA…
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
(T = MOMENTARY SWITCH IS TOGGLED; 0 = OPEN; 1 = CLOSED)
MS2
0
0
0
1
1
1
0
0
0
1
0
0
0
0
0
0
MS1
0
T
T
0
T
T
0
T
T
0
0
T
T
T
T
T
Memory
A
B
C
D
D
D
C
A
B
D
B
C
A
B
C
A
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RHYTHM R3910
Mode 3: Static Switch on MS1 and MS2
When MS2 is set CLOSED, the state of the switch on MS1
is ignored. This prevents memory select beeps from
occurring if switching MS1 when MS2 is CLOSED. The
part starts in whatever memory the switches are selecting. If
a memory is invalid, the part defaults to memory A. The part
starts in whatever memory the switches are selecting. If
a memory is invalid, the part defaults to memory A.
This mode uses two static switches to change memories.
Table 8 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’.
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 R3910 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.
R3910 also includes the Peak Clipper block for added
flexibility.
Table 8. MEMORY SELECTED BY STATIC SWITCH
ON MS1 AND MS2 MODE; (EXAMPLE WITH FOUR
VALID MEMORIES)
MS1
MS2
Memory
OPEN
OPEN
A
CLOSED
OPEN
B (if valid, otherwise A)
OPEN
CLOSED
C (if valid, otherwise A)
CLOSED
CLOSED
D (if valid, otherwise A)
Mode 4: 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 OPEN
or CLOSED depending on the configuration of MS2. This
means that this mode will only use a maximum of three
memories (even if four valid memories are programmed).
Tables 9 describes 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’.
Memory Switch Fader
To minimize potential loud transients when switching
between memories, R3910 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.
Table 9. MEMORY SELECTED BY STATIC SWITCH
ON MS1, STATIC SWITCH ON MS2 (JUMP TO LAST
MEMORY) MODE
MS1
MS2
Memory
OPEN
OPEN
A
CLOSED
OPEN
B (if valid, otherwise A)
OPEN
CLOSED
D
CLOSED
CLOSED
D
Power Management
R3910 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.
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.
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
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RHYTHM R3910
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.
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.
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.
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.
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.
Advanced Reset Mode
Software and Security
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.
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
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17
RHYTHM R3910
Power Supply Considerations
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.
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.
SDA and I2C Communication
Input Connection and Layout Considerations
R3910 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
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. I2C 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.
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.
•
ORDERING INFORMATION
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
Device
†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.
www.onsemi.com
18
RHYTHM R3910
PAD LOCATIONS
Table 10. PAD POSITION AND DIMENSIONS
Pad Position
Pad Dimensions
Pad No.
X
Y
Xdim (mil)
Ydim (mil)
1
0
0
20
33
2
−27
0
20
33
3
−54
−5
20
23
4
−81
−5
20
23
5
−108
−5
20
23
6
−135
−5
20
23
7
−162
−5
20
23
8
−189
0
20
33
9
−189
42
20
23
10
−189
85
20
23
11
−162
85
20
23
12
−135
85
20
23
13
−108
85
20
23
14
−81
85
20
23
15
−54
85
20
23
16
−27
85
20
23
17
0
85
20
23
18
0
42
20
23
19
−27
42
20
23
20
−54
42
20
23
21
−81
42
20
23
22
−108
42
20
23
23
−135
42
20
23
24
−162
26.5
18
12
25
−162
53.5
18
12
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19
RHYTHM R3910
Table 10. PAD POSITION AND DIMENSIONS
Pad No.
X
Y
Xdim (mm)
Ydim (mm)
1
0
0
0.508
0.838
2
−0.686
0
0.508
0.838
3
−1.372
−0.127
0.508
0.584
4
−2.057
−0.127
0.508
0.584
5
−2.743
−0.127
0.508
0.584
6
−3.429
−0.127
0.508
0.584
7
−4.115
−0.127
0.508
0.584
8
−4.801
0
0.508
0.838
9
−4.801
1.067
0.508
0.584
10
−4.801
2.159
0.508
0.584
11
−4.115
2.159
0.508
0.584
12
−3.429
2.159
0.508
0.584
13
−2.743
2.159
0.508
0.584
14
−2.057
2.159
0.508
0.584
15
−1.372
2.159
0.508
0.584
16
−0.686
2.159
0.508
0.584
17
0
2.159
0.508
0.584
18
0
1.067
0.508
0.584
19
−0.686
1.067
0.508
0.584
20
−1.372
1.067
0.508
0.584
21
−2.057
1.067
0.508
0.584
22
−2.743
1.067
0.508
0.584
23
−3.429
1.067
0.508
0.584
24
−4.115
0.673
0.457
0.305
25
−4.115
1.359
0.457
0.305
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20
RHYTHM R3910
PACKAGE DIMENSIONS
SIP25, 5.59x3.18
CASE 127DN
ISSUE O
E
NOTES:
1. DIMENSIONING AND TOLERANCING PER
ASME Y14.5M, 1994.
2. CONTROLLING DIMENSION: MILLIMETERS.
3. COPLANARITY APPLIES TO THE SPHERICAL
CROWNS OF THE PADS.
A B
D
PIN A1
INDICATOR
0.13 C
2X
0.13 C
2X
e
ÈÈÈ
ÈÈÈ
3X
L2
20X
2X
TOP VIEW
L
L3
A2
2X
b1
0.13 C
DETAIL A
A
0.05 C
A1
NOTE 3
C
SIDE VIEW
DIM
A
A1
A2
b
b1
D
E
e
e1
e2
e3
L
L2
L3
MILLIMETERS
MIN
MAX
−−−
1.83
0.08
0.18
−−−
1.65
0.478
0.538
0.427
0.487
3.18 BSC
5.59 BSC
0.686 BSC
0.051 BSC
1.067 BSC
1.092 BSC
0.554
0.614
0.808
0.868
0.275
0.335
SEATING
PLANE
e
e/2
e2
A
e
e/2
B
e1
C
e3
1
2
3
4
5
6
7
8
23X
b
0.05
0.03
BOTTOM VIEW
C A
C
B
RECOMMENDED
SOLDERING FOOTPRINT*
NOTE 4
23X
0.538
0.686
1.092
20X
1.067
0.614
2X
2X
0.487
0.335
0.051
3X
0.686
0.868
A1
DETAIL B
0.686
PITCH
DETAIL B
DIMENSIONS: MILLIMETERS
*For additional information on our Pb−Free strategy and soldering
details, please download the ON Semiconductor Soldering and
Mounting Techniques Reference Manual, SOLDERRM/D.
www.onsemi.com
21
RHYTHM R3910
iSceneDetect, iLog, RHYTHM, HRX and EVOKE are trademarks of Semiconductor Components Industries, LLC.
thinSTAX, FRONTWAVE, and ARKonline are registered trademarks of Semiconductor Components Industries, LLC.
ON Semiconductor and the
are registered trademarks of Semiconductor Components Industries, LLC (SCILLC) or its subsidiaries 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
at www.onsemi.com/site/pdf/Patent−Marking.pdf. SCILLC reserves the right to make changes without further notice to any products herein. SCILLC makes no warranty, representation
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
LITERATURE FULFILLMENT:
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22
ON Semiconductor Website: www.onsemi.com
Order Literature: http://www.onsemi.com/orderlit
For additional information, please contact your local
Sales Representative
R3910/D