Advance acoustic Terra 128B Specifications

RHYTHM R3710
Preconfigured DSP System
Description
RHYTHMt R3710 is a preconfigured DSP system designed
specifically for Invisible−In−Canal (IIC) hearing aid devices.
Available in the industry’s smallest hybrid form−factors, it is well
suited for hearing aid designs that are placed deep in the ear canal.
Using miniaturized advanced packaging techniques,
ON Semiconductor enables hearing aid manufacturers to take
advantage of the highly compact size to produce IIC hearing aids that
fit a greater portion of the market. Featuring iSceneDetectt
environmental classification, adaptive noise reduction, feedback
cancellation, and up to 8−channel WDRC, R3710 provides the
advanced features and performance level typically found in high−end
products.
Acoustic Environment Classification − The iSceneDetect
environmental classification algorithm is capable of analyzing the
hearing aid user’s acoustic environment and automatically optimizes
the hearing aid to maximize comfort and audibility.
iLogt 6.0 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 during follow up visits.
Evoket Advanced Acoustic Indicators − Allows manufacturers
to provide more pleasing, multi−frequency tones simulating musical
notes or chords to indicate events such as program or volume changes.
Adaptive Feedback Canceller − Automatically reduces acoustic
feedback and allows for an increase in the stable gain while
minimizing artifacts for music and tonal input signals.
Adaptive Noise Reduction − The adaptive noise reduction
algorithm on R3710 monitors noise levels independently in 128
individual bands and employs advanced psychoacoustic models to
provide user comfort.
Tinnitus Masking − R3710 is equipped with a noise source that can
be used to mask tinnitus. The noise can be shaped, attenuated, and duty
cycled then summed into the audio path either before or after the
volume control.
In−situ Tone and Noise Generator − The narrow−band noise and
tone 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.
© Semiconductor Components Industries, LLC, 2014
April, 2013 − Rev. 4
1
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PAD CONNECTION
MGND
VREG
1
VIN2
13
2
VIN1
MS1
12
3
MS2
SDA
11
4
DVC
CLK
10
5
OUT+
VB
9
6
OUT−
NC
14
VC
15
8
GND2
16
7
GND1
(Bottom View)
MARKING DIAGRAM
R3710
XXXXXX
R3710
= Specific Device Code
XXXXXX = Work Order Number
ORDERING INFORMATION
See detailed ordering and shipping information on page 15 of
this data sheet.
Publication Order Number:
R3710/D
RHYTHM R3710
• 2 Analog Inputs
• 16 kHz or 8 kHz Bandwidth
• 4 Fully Configurable Memories with Audible Memory
Other Key Features − R3710 also supports the following
features: 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. R3710 also encompasses
industry−leading security features to avoid cloning and
software piracy.
•
•
•
•
•
•
Features
• Advanced Research Algorithms:
♦
iSceneDetect Environmental Classification
128−band Adaptive Noise Reduction
♦ Adaptive Feedback Cancellation (AFC)
iLog 6.0 Datalogging
Tinnitus Masking Noise Generator
Evoke Acoustic Indicators
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
♦
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
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, Halogen Free/BFR Free
and are RoHS Compliant
thinSTAX Packaging
• Hybrid Typical Dimensions:
0.180 x 0.123 x 0.060 in. (nominal)
(4.57 x 3.12 x 1.52 mm)
BLOCK DIAGRAM
VB
9
RHYTHM R3710
VREG 16
VIN1
2
VIN2
1
ACOUSTIC
INDICATORS
VOLTAGE
REGULATOR
POST
BIQUAD
FILTERS
3&4
CROSS
FADER
S
FEEDBACK
CANCELLER
M
U
X
POR
CIRCUITRY
PEAK
CLIPPER
D/A
HBRIDGE
5 OUT+
CONTROL
A/D
13 VC
NOISE GENERATOR
AND SHAPER
A/D
VOLUME
CONTROL
AGCO
PRE
BI Q UA D
FILTERS
WIDEBAND
GAIN
S
POST
BIQUAD
FILTERS
1&2
ENVIRONMENTAL
CLASSIFICATION
4 DVC
MGND 15
FREQUENCY
BA N D
ANALYSIS
SDA 11
CLK 10
6 OUT−
128 bands
FREQUENCY
BA N D
SYNTHESIS
CONTROL
(MS/DIGVC)
3 MS2
WDRC (1,2,4,6 or 8 channels)
PROGRAMMING
INTERFACE
Noise Reduction (128 bands)
DATA
LOGGING
Graphic EQ (16 bands)
CLOCK
GENERATOR
14
7
8
NC
GND1
GND2
Figure 1. Hybrid Block Diagram
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12 MS1
EEPROM
RHYTHM R3710
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
Supply Voltage Turn On Threshold
VBON
Ramp up, zinc−air
1.06
1.10
1.16
Ramp up, NiMH
1.16
1.20
1.24
All functions, 32 kHz sampling rate
−
665
−
All functions, 16 kHz sampling rate
−
575
−
Hybrid Current
V
mA
EEPROM Burn Cycles
−
−
100 k
−
−
cycles
Low Frequency System Limit
−
−
−
125
−
Hz
High Frequency System Limit
−
−
−
16
−
kHz
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 VIN1 and VIN2
−
60
−
dB
Total Harmonic Distortion
THD at Maximum Input
Clock Frequency
REGULATOR
Regulator Voltage
System PSRR
INPUT
Maximum Input Level
−
−
−
−15
−13
dBV
VAN_IN
VIN1, VIN2
0
−
800
mV
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
−
−
7
−
−
bits
Analogue Input Voltage Range
OUTPUT
D/A Dynamic Range
Output Impedance
CONTROL A/D
Resolution (monotonic)
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RHYTHM R3710
Table 2. ELECTRICAL CHARACTERISTICS (Supply Voltage VB = 1.25 V; Temperature = 25°C) (continued)
Parameter
Symbol
Conditions
Min
Typ
Max
Units
Zero Scale Level
−
−
−
0
−
V
Full Scale Level
−
−
−
VREG
−
V
RVC
Three−terminal connection
100
−
360
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 R3710
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 R3710
Figure 2. I2C Mode Timing
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RHYTHM R3710
TYPICAL APPLICATIONS
9
RHYTHM R3710
ACOUSTIC
INDICATORS
OUT
VOLTAGE
REGULATOR
16
POST
BIQUAD
FILTERS
3&4
CROSS
FADER
S
FEEDBACK
CANCELLER
3k75
2
M
U
X
3k75
1
POR
CIRCUITRY
PEAK
CLIPPER
CONTROL
A/D
13
CONTROL
(MS/DIGVC)
12
LP FILTER
6
NOISE GENERATOR
AND SHAPER
VOLUME
CONTROL
AGCO
WIDEBAND
GAIN
S
FREQUENCY
BA N D
ANALYSIS
200k
POST
BIQUAD
FILTERS
1&2
ENVIRONMENTAL
CLASSIFICATION
15
128 bands
4
FREQUENCY
BA N D
SYNTHESIS
3
WDRC (1,2,4,6 or 8 channels)
PROGRAMMING
INTERFACE
10
5
A/D
PRE
BI Q UA D
FILTERS
11
D/A
HBRIDGE
Noise Reduction (128 bands)
DATA
LOGGING
Graphic EQ (16 bands)
EEPROM
CLOCK
GENERATOR
14
8
7
Figure 3. Test Circuit
9
RHYTHM R3710
16
2
Microphone
1
ACOUSTIC
INDICATORS
VOLTAGE
REGULATOR
POST
BIQUAD
FILTERS
3&4
CROSS
FADER
S
FEEDBACK
CANCELLER
M
U
X
POR
CIRCUITRY
PEAK
CLIPPER
D/A
HBRIDGE
5
Zero−bias
6
Receiver
CONTROL
A/D
13
NOISE GENERATOR
AND SHAPER
A/D
AGCO
PRE
BI Q UA D
FILTERS
VOLUME
CONTROL
S
WIDEBAND
GAIN
POST
BIQUAD
FILTERS
1&2
ENVIRONMENTAL
CLASSIFICATION
4
15
FREQUENCY
BA N D
ANALYSIS
11
10
128 bands
FREQUENCY
BA N D
SYNTHESIS
CONTROL
(MS/DIGVC)
3
WDRC (1,2,4,6 or 8 channels)
PROGRAMMING
INTERFACE
Noise Reduction (128 bands)
DATA
LOGGING
Graphic EQ (16 bands)
CLOCK
GENERATOR
14
7
Figure 4. Typical Application Circuit
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8
12
EEPROM
RHYTHM R3710
SIGNAL PATH
Evoke Advanced Acoustic Indicators
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.
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.
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
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.
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.
Adaptive Feedback Canceller
The Adaptive Feedback Canceller (AFC) reduces
acoustic feedback by forming an estimate of the hearing aid
feedback signal and then subtracting this estimate from the
hearing aid input. The forward path of the hearing aid is not
affected. Unlike adaptive notch filter approaches, the AFC
algorithm 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.
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, 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 the
specific environment.
iSceneDetect will gradually make the adjustments so the
change in settings based on the environment is smooth and
virtually unnoticeable. This feature will enable the hearing
aid wearer to have an instrument which will work in any
environment with a single memory.
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RHYTHM R3710
Feedback path
+
−
Σ
In−Situ Datalogging − iLog 6.0
H
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
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.
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, 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.
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.
Tinnitus Treatment
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
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RHYTHM R3710
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.
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).
HRX Head Room Expander
R3710 has an enhanced Head Room Extension (HRX)
circuit that increases the input dynamic range of 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)
Table 4. NOISE INJECTION EFFECT ON VC
Off
0
VC Controls
Noise
Injected
Audio
Enabled
Audio
Off
Yes
Pre VC
Audio + Noise
Pre VC
Yes
Post VC
Audio
Post VC
Yes
Noise only
Pre VC
Noise
Pre VC
No
Noise only
Post VC
−
Post VC
No
Pre VC with
Noise
Noise
Pre VC
Yes
High Level
Gain
−10
OUTPUT LEVEL (dBV)
Noise
Insertion
Modes
−20
−30 Low Level
−40 Gain
−50
Compression
Ratio
Lower
Threshold
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
Narrow−band Tone and Noise Stimulus
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.
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)
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
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
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RHYTHM R3710
Volume Control and Switches
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.
External Volume Control
The volume of the device can either be set statically via
software or controlled externally via a physical interface.
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.
Graphic Equalizer
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.
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.
Biquad Filters
Additional frequency shaping can be achieved by
configuring generic biquad filters. The transfer function for
each of the biquad filters is as follows:
H(z) + b0 ) b1
1 ) a1
z −1 ) b2
z −1 ) a2
z −2
z −2
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 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.
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
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.
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 R3710. This enables user’s tremendous
flexibility in switching between configurations. Up to four
memories can be configured and selected by the MS
switches on R3710. 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.
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RHYTHM R3710
• Donly: this parameter determines whether the MS2
The behavior of the MS switches is controlled by two main
parameters in IDS.
• MSSmode: this mode determines whether a connected
switch is momentary or static.
switch is dedicated to the last memory position.
There are four basic MS switch modes of operation as shown
in Table 5 below.
Table 5. MS SWITCH MODES
MS Switch Mode
MS1 Switch
Max # of valid
Memories
MS2 Switch
Donly
MSSMode
Use
Mode 1
Momentary
None
4
Off
Momentary
Simplest configuration
Mode 2
Momentary
Static
4
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
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
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. Startup or during normal operation.
If the static switch on MS2 is CLOSED, 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 MS2 = CLOSED: D, then memory D is enabled
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.
Mode 1: 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…
Mode 2: 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
Table 7. DYNAMIC EXAMPLE WITH FOUR VALID MEMORIES AND MS2 PULL−UP/PULL−DOWN = PULL−DOWN
(T = MOMENTARY SWITCH IS TOGGLED; 0 = OPEN; 1 = HIGH)
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 R3710
Mode 3: Static Switch on MS1 and MS2
When MS2 is 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 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.
R3710 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
CLOSED. This means that this mode will only use a
maximum of three memories (even if four valid memories
are programmed). Table 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, 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.
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
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.
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
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RHYTHM R3710
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.
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.
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 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
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
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RHYTHM R3710
•
Power Supply Considerations
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.
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. 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.
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.
Input Connection and Layout Considerations
It is recommended to connect unused audio input pins
directly to MGND to minimize the possibility of noise
pickup. Inputs are internally AC coupled, so there is no
additional leakage current when inputs are connected
directly to ground.
In order to further minimize noise at the inputs the
following guidelines are recommended:
• MGND is used as reference ground plane for input
signals. All input components should be grounded to
MGND. This ground plane should be isolated from all
other ground connections in the system.
• Keep the input traces as short as possible and avoid
routing traces near high noise sources such as the
OUT+ and OUT− pins
• Star ground input component grounds to the MGND
connection.
SDA and I2C Communication
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
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, 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. I2C programming
support is available on the HiPro (serial or USB versions)
and ON Semiconductor’s DSP Programmer 3.0.
ORDERING INFORMATION
Package
Shipping†
R3710−CEAA−E1T
25 Pad Hybrid
250 Units / Tape & Reel
R3710−CEAA−E1
25 Pad Hybrid
25 Units / Bubble Pack
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 R3710 contact your Sales Account Manager or FAE and use part number
R3710GEVB.
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RHYTHM R3710
Table 10. PAD POSITION AND DIMENSIONS (mil)
Pad Position
Pad Dimensions
Pad No.
Pad Name
X
Y
Xdim
Ydim
1
VIN2
−71
44
23
23
2
VIN1
−42
44
21
23
3
MS2
−14
44
21
23
4
DVC
14
44
21
23
5
OUT+
42
44
21
23
6
OUT−
71
44
23
23
7
GND1
71
14.5
23
21
8
GND2
71
−14.5
23
21
9
VB
71
−44
23
23
10
CLK
42
−44
21
23
11
SDA
14
−44
21
23
12
MS1
−14
−44
21
23
13
VC
−42
−44
21
23
14
NC
−71
−44
23
23
15
MGND
−71
−14.5
23
21
16
VREG
−71
14.5
23
21
5. Pin location is referenced to the center of the hybrid device.
6. Pad position is relative to the center of the hybrid pad.
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RHYTHM R3710
Table 11. PAD POSITION AND DIMENSIONS (mm)
Pad Position
Pad Dimensions
Pad No.
Pad Name
X
Y
Xdim
Ydim
1
VIN2
−1.8034
1.1176
0.5842
0.5842
2
VIN1
−1.0668
1.1176
0.5334
0.5842
3
MS2
−0.3556
1.1176
0.5334
0.5842
4
DVC
0.3556
1.1176
0.5334
0.5842
5
OUT+
1.0668
1.1176
0.5334
0.5842
6
OUT−
1.8034
1.1176
0.5842
0.5842
7
GND1
1.8034
0.3683
0.5842
0.5334
8
GND2
1.8034
−0.3683
0.5842
0.5334
9
VB
1.8034
−1.1176
0.5842
0.5842
10
CLK
1.0668
−1.1176
0.5334
0.5842
11
SDA
0.3556
−1.1176
0.5334
0.5842
12
MS1
−0.3556
−1.1176
0.5334
0.5842
13
VC
−1.0668
−1.1176
0.5334
0.5842
14
NC
−1.8034
−1.1176
0.5842
0.5842
15
MGND
−1.8034
−0.3683
0.5842
0.5334
16
VREG
−1.8034
0.3683
0.5842
0.5334
7. Pin location is referenced to the center of the hybrid device.
8. Pad position is relative to the center of the hybrid pad.
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RHYTHM R3710
PACKAGE DIMENSIONS
4.57mm±0.13mm
(180mil±5mil)
3.12mm±0.13mm
(123mil±5mil)
R3710−CEAA
XXXXXX
1.39mm MAX
(55mil MAX)
0.08mm ±0.05mm
(3mil ±2mil)
PIN 1 INDICATOR
Dimension units are in inches.
Dimensions in parentheses are in millimeters, converted from inches and include minor rounding errors.
1.000 inches = 25.4 mm
Hybrid Dimension Tolerances: ±0.005 (±0.13)
Solder Pad Height Tolerances: ±0.002 (±0.05)
= location of Pin 1
E1: RoHS compliant hybrid, MSL#4, 240°C peak reflow, SAC305
This Hybrid is designed for either point−to−point manual soldering or for reflow according to ON Semiconductor’s reflow process.
iSceneDetect, iLog, HRX, RHYTHM, WOLVERINE, thinSTAX and EVOKE are trademarks of Semiconductor Components Industries, LLC.
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