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PDF SA3291 Data sheet ( Hoja de datos )
| Número de pieza | SA3291 | |
| Descripción | Preconfigured Wireless DSP System | |
| Fabricantes | ON Semiconductor | |
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AYRE SA3291
Preconfigured Wireless DSP
System for Hearing Aids
Description
Ayret SA3291 is a preconfigured wireless DSP system utilizing
Near−Field Magnetic Induction (NFMI) technology.
Ayre SA3291 enables hearing aids to wirelessly synchronize
program modes and volume control and stream telecoil signals from
one hearing aid to the other. When coupled with a relay device, Ayre
SA3291 enables features such as stereo audio streaming via
Bluetooth® in addition to remote control functionality.
Featuring iSceneDetectt environmental classification, adaptive
noise reduction, superior feedback cancellation, fully automated and
adaptive microphone directionality, and up to 8−channel WDRC, Ayre
SA3291 is ideal for high−end, full−featured wireless products.
Binaural Synchronization − The binaural synchronization feature
allows two hearing aids to wirelessly synchronize adjustments such as
program modes or volume control. By working together as one
system, user control is simplified.
Binaural Telecoil − The binaural telecoil wirelessly streams telecoil
audio signals from one hearing aid to the other. This enables hearing
aid users to hear phone calls in both ears, improving speech
intelligibility.
Stereo Audio Streaming − Stereo audio signals can be streamed
wirelessly from a relay device to hearing aids equipped with Ayre
SA3291. A relay device can use Bluetooth or other far−field wireless
technology to wirelessly connect with TVs, music players, mobile
phones or other audio sources.
Acoustic Environment Classification − The iSceneDetect 1.0
environmental classification algorithm is capable of analyzing the
hearing aid wearer’s acoustic environment and automatically
optimizes the hearing aid to maximize comfort and audibility.
iLog] 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[ 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 Cancellation − Automatically reduces
acoustic feedback. It allows for an increase in the stable gain while
minimizing artifacts for music and tonal input signals.
www.onsemi.com
32 PAD
HYBRID
CASE TBD
GND
DAI
VB
GPIO0/
MS1
GPIO1/
MS2
PCSDA
PCCLK
OUT−
PAD CONNECTION
MGND VIN2
20
21
22
19 27
1
2
18 28 3
17 23
4
29
16 24
5
30
15 25
31
14 26
32
13
6
7
8
VIN1
TIN
VREG
VC
GPIO7/
DIGVC
L2
NGND
L1
OUT+ 12 11 10
9 NVB
VBP PGND
(Bottom View)
MARKING DIAGRAM
SA3291A−E1
XXXXXX
SA3291A = Specific Device Code
E1 = RoHS Compliant Hybrid
XXXXXX = Work Order Number
ORDERING INFORMATION
See detailed ordering and shipping information on page 17 of
this data sheet.
© Semiconductor Components Industries, LLC, 2015
March, 2015 − Rev. 7
1
Publication Order Number:
SA3291/D
1 page  
AYRE SA3291
Table 2. ELECTRICAL CHARACTERISTICS (Supply Voltage VB = NVB = 1.25 V; Temperature = 25°C) (continued)
Parameter
Symbol
Conditions
Min Typ Max
Units
INPUT
Analog Input Voltage Range
Input Dynamic Range
VAN_IN
VAN_TIN
−
VIN1, VIN2, Al
TIN
HRX − ON Bandwidth
100 Hz − 8 kHz
0
−100
−
−
−
95
800 mV
800
96 dB
Audio Sampling Rate
−
− 8 − 48 kHz
OUTPUT
D/A Dynamic Range
−
100 Hz − 8 kHz
− 88 − dB
Output Impedance
VOLUME CONTROL
ZOUT
−
− 10 13 W
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 
AYRE SA3291
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 (ADM)
ON Semiconductor’s Adaptive Directional Microphone
algorithm 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
The Ayre SA3291 has a datalogging function that records
information every 4 s 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
• Wireless audio and phone streaming
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.
www.onsemi.com
11
11 Page | ||
| Páginas | Total 20 Páginas | |
| PDF Descargar | [ Datasheet SA3291.PDF ] | |
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