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PDF R3910 Data sheet ( Hoja de datos )
| Número de pieza | R3910 | |
| Descripción | Preconfigured DSP System | |
| Fabricantes | ON Semiconductor | |
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RHYTHM R3910
Preconfigured DSP System
for Hearing Aids
Description
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.
EVOKE t Advanced Acoustic Indicators − Allows
manufacturers to provide more pleasing, multi−frequency tones
simulating musical notes or chords to indicate events such as program
or volume changes.
Automatic Adaptive Directionality − The automatic Adaptive
Directional Microphone (ADM) algorithm automatically reduces the
level of sound sources that originate from behind or to the side of the
hearing aid wearer without affecting sounds from the front. The
algorithm can also gather input from the acoustic environment and
automatically select whether directionality is needed or not,
translating into additional current savings.
Adaptive Feedback Canceller − Automatically reduces acoustic
feedback. It allows for an increase in the stable gain while minimizing
artifacts for music and tonal input signals.
Adaptive Noise Reduction − The adaptive noise algorithm on
R3910 monitors noise levels independently in 128 individual bands
and employs advanced 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.
www.onsemi.com
25 PAD
HYBRID
CASE 127DN
PAD CONNECTION
VIN2 17
TIN 16
DAI 15
VC 14
D_VC 13
SDA 12
CLK 11
18
VIN1
19
N/C
20
N/C
21
N/C
22
N/C
23
N/C
25 24
1 VREG
2 MGND
3 GND
4 PGND
5 OUT+
6 OUT−
7 VBP
MS1 10
9
8
MS2
(Bottom View)
MARKING DIAGRAM
VB
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.
© Semiconductor Components Industries, LLC, 2015
August, 2015 − Rev. 9
1
Publication Order Number:
R3910/D
1 page  
RHYTHM R3910
Table 2. ELECTRICAL CHARACTERISTICS (Supply Voltage VB = 1.25 V; Temperature = 25°C) (continued)
Parameter
Symbol
Conditions
Min Typ
Max
Units
CONTROL A/D
Resolution (monotonic)
−
− 7 − − bits
Zero Scale Level
− − −0−V
Full Scale Level
VOLUME CONTROL
−
−
− VREG −
V
Volume Control Resistance
Volume Control Range
RVC
Three−terminal connection
200
−
1000
kW
− − − − 42 dB
PC_SDA INPUT
Logic 0 Voltage
− − 0 − 0.3 V
Logic 1 Voltage
− − 1 − 1.25 V
PC_SDA OUTPUT
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
Synchronization Time
(Synchronization Pulse Width)
TSYNC
Baud = 0
Baud = 1
237 250 263
118 125 132
ms
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
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.
www.onsemi.com
5
5 Page 
RHYTHM R3910
The noise reduction gain applied to a given band is
determined by a combination of three factors:
• Signal−to−Noise Ratio (SNR)
• Masking threshold
• Dynamics of the SNR per band
The SNR in each band determines the maximum amount
of attenuation to be applied to the band − the poorer the SNR,
the greater the amount of attenuation. Simultaneously, in
each band, the masking threshold variations resulting from
the energy in other adjacent bands is taken into account.
Finally, the noise reduction gain is also adjusted to take
advantage of the natural masking of ‘noisy’ bands by speech
bands over time.
Based on this approach, only enough attenuation is
applied to bring the energy in each ‘noisy’ band to just below
the masking threshold. This prevents excessive amounts of
attenuation from being applied and thereby reduces
unwanted artifacts and audio distortion. The Noise
Reduction algorithm efficiently removes a wide variety of
types of noise, while retaining natural speech quality and
level. The level of noise reduction (aggressiveness) is
configurable to 3, 6, 9 and 12 dB of reduction.
Directional Microphones
In any directional mode, the circuitry includes a fixed
filter for compensating the sensitivity and frequency
response differences between microphones. The filter
parameters are adjusted during product calibration.
A dedicated biquad filter following the directional block
has been allocated for low frequency equalization to
compensate for the 6 dB/octave roll−off in frequency
response that occurs in directional mode. The amount of low
frequency equalization that is applied is programmable.
ON Semiconductor recommends using matched
microphones. The maximum spacing between the front and
rear microphones cannot exceed 20 mm (0.787 in).
Adaptive Directional Microphones
The Adaptive Directional Microphone (ADM) algorithm
from ON Semiconductor is a two−microphone processing
scheme for hearing aids. It is designed to automatically
reduce the level of sound sources that originate from behind
or the side of the hearing−aid wearer without affecting
sounds from the front. The algorithm accomplishes this by
adjusting the null in the microphone polar pattern to
minimize the noise level at the output of the ADM. The
discrimination between desired signal and noise is based
entirely on the direction of arrival with respect to the hearing
aid: sounds from the front hemisphere are passed
unattenuated whereas sounds arriving from the rear
hemisphere are reduced.
The angular location of the null in the microphone polar
pattern is continuously variable over a range of 90 to 180
degrees where 0 degrees represents the front.
The location of the null in the microphone pattern is
influenced by the nature of the acoustic signals (spectral
content, direction of arrival) as well as the acoustical
characteristics of the room. The ADM algorithm steers a
single, broadband null to a location that minimizes the
output noise power. If a specific noise signal has frequency
components that are dominant, then these will have a larger
influence on the null location than a weaker signal at a
different location. In addition, the position of the null is
affected by acoustic reflections. The presence of an acoustic
reflection may cause a noise source to appear as if it
originates at a location other than the true location. In this
case, the ADM algorithm chooses a compromise null
location that minimizes the level of noise at the ADM
output.
Automatic Adaptive Directional Microphones
When Automatic ADM mode is selected, the adaptive
directional microphone remains enabled as long as the
ambient sound level is above a specific threshold and the
directional microphone has not converged to an
omni−directional polar pattern. On the other hand, if the
ambient sound level is below a specific threshold, or if the
directional microphone has converged to an omni−directional
polar pattern, then the algorithm will switch to single
microphone, omni−directional state to reduce current
consumption. While in this omni−directional state, the
algorithm will periodically check for conditions warranting
the enabling of the adaptive directional microphone.
FrontWave Directionality
The FrontWave block provides the resources necessary to
implement directional microphone processing. The block
accepts inputs from both a front and rear microphone and
provides a synthesized directional microphone signal as its
output. The directional microphone output is obtained by
delaying the rear microphone signal and subtracting it from
the front microphone signal. Various microphone response
patterns can be obtained by adjusting the time delay.
In−Situ Datalogging − iLog 4.0
R3910 has a datalogging function that records
information every 4 seconds to 60 minutes (programmable)
about the state of the hearing aid and its environment to
non−volatile memory. The function can be enabled with the
ARK software and information collection will begin the
next time the hybrid is powered up. This information is
recorded over time and can be downloaded for analysis.
The following parameters are sampled:
• Battery level
• Volume control setting
• Program memory selection
• Environment
• Ambient sound level
• Length of time the hearing aid was powered on
www.onsemi.com
11
11 Page | ||
| Páginas | Total 22 Páginas | |
| PDF Descargar | [ Datasheet R3910.PDF ] | |
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