Preliminary

AS2520/21/20B/21B

Rev. 5.1

Page 1

May 1999

Key Features

Line/speech circuit, loudhearing, handsfree and
dc/dc converter on one 28 pin CMOS chip

Operating range from 13 to 100 mA (down to 5
mA with reduced performance)

Soft clipping control eliminating harsh distortion

Volume control of receive signal with squelch and
automatic loop gain compensation

Line loss compensation pin selectable

Low noise (max. - 72 dBmp)

Real or complex impedance adjustable

NET 4 compatible

Dynamically controlled voice switching

Same monitor amplifier for loudhearing, handsfree
and tone ringer

Very few external components

Power derived from ring signal by switching

converter during ringing

Typical Application

General Description

The AS2520/21/20B/21B are CMOS integrated
circuits that contain all the audio functions needed to
form a high comfort, line-powered telephone.

The devices incorporate line adaptation, speech
circuit, loudhearing and handsfree - all supervised by
the novel voice and power control circuit. A switching
converter is also provided for converting the ring
signal. The interface to a dialler/controller is made
very simple to allow easy adaptation to a telecom
microcontroller.

The AS2520 series incorporate volume control for the
earpiece and the loudspeaker (AS2520 digital with +/-
keys and AS2521 analogue with potentiometer). The
volume control circuit automatically compensates the
loop gain to ensure acoustic stability.

Package

Available in 28 pin SOP and DIP.

AS2520

DIALLER

礐ONTROLLER

LCD DRIVER

HSM

HFM

TELEPHONE

SPEECH CIRCUIT

WITH

LOUDHEARING,

HANDSFREE,

DC/DC CONVERTER

Figure 1: Typical Handsfree Telephone Application

Austria Mikro Systeme International AG

Telephone Speech Circuit

with Loudhearing and Handsfree

Preliminary

AS2520/21/20B/21B

Rev. 5.1

Page 2

May 1999

Pin Description

Pin #

Name

Type

Description

Line Current Sense Input
This input is used for sensing the line current.

Complex Impedance Input

Input pin for the capacitor in the complex impedance.

Receive Output
This is the output for driving a dynamic earpiece with an impedance of 140
to 300 ohm.

Supply

Positive Voltage Supply
This is the supply pin for the circuit.

GND

Supply

Analogue Ground
This pin is the analogue ground for the amplifiers.

STB

Side Tone Balance Input
This is the input for the side tone cancellation network.

LLC

Line Loss Compensation Selection Pin
LLC = V

High range

-6 dB from 45 mA to 75 mA;

LLC = A

GND

: Low range

-6 dB from 20 mA to 50 mA;

LLC = V

: No regulation

gain independent of line current;

LSI

Loudspeaker Amplifier Input
This is the input for applying the receive signal to the loudspeaker
amplifier.

AI
DI

Tone Input
This switchable input is intended for transmitting DTMF or other signals
like messages on TAMs (Telephone Answering Machines) onto the line in
off-hook conditions and when in ringing mode to apply a PDM signal to the
loudspeaker (see also table 1).

RTH

Receive Threshold Input
The sensibility of the receive peak detector can be adjusted by applying
the signal from RO to the RTH input through a voltage divider.

Converter Make Output
This is an output for controlling the external switching converter. It
converts the ring signal into a 4V supply voltage and is activated when PD
= high and HS, LE, MT = low.

Supply

Loudspeaker Power Supply
High power supply for the output driver stage.

Output for Loudspeaker
Output pin for an ac coupled 32

(25 to 50

)loudspeaker.

SSP

Supply

Negative High Power Supply
This pin is the negative high power supply for the loudspeaker amplifier.

Mute Input
Dialling mute input (see also table 1).
MT = V

: Tx and Rx channels muted;

MT = V

: Tx and Rx channels not muted.

Power Down Input
Input for powering down the speech circuit and loudhearing/handsfree
(see table 1).

Loudhearing Enable Input
Input for enabling loudhearing/handsfree, active high (see table 1).

Handset Switch Input
This is an input that is pulled high by the hook switch (handset) or 礐
when off-hook (see table 1).

Preliminary

AS2520/21/20B/21B

Rev. 5.1

Page 3

May 1999

19
22

M1
M2

Microphone Inputs
Differential inputs for handset microphone (electret).

20
21

M4
M3

Handsfree Microphone Inputs
These are the input pins for the handsfree microphone (electret).

VOL

D/AI

Volume Control Input
Volume control for the receive signal.
AS2520: Digital control with +/� keys or from 礐;
AS2521: Analogue dc control with potentiometer.

Supply Source Control Output
This N-channel open drain output controls the external high power source
transistor for supplying (V

) the loudspeaker amplifier in off-hook

loudhearing/handsfree mode.

Current Shunt Control Output
This N-channel open drain output controls the external high power shunt
transistor for the modulation of the line voltage and for shorting the line
during make period of pulse dialling.

Supply

Negative Power Supply

AI/O

Line Input
This input is used for power extraction and line current sensing.

Receive Input
This is the input for the receive signal.

DI:

Digital Input

AI:

Analogue Input

DO:

Digital Output

AO:

Analogue Output

DI/O:

Digital Input/Output

AI/O:

Analogue Input/output

Operating Modes

I/O Pins

Digital Inputs

Tone Input

Outputs

MODE

Idle (on-hook)

Not connected

Low

Ringing

PDM signal to LO (DI)

`TI'

POT

Not connected

Low

`M1/M2'

`RI/STB'

POT/pulse dialling

Not connected

Low

POT/DTMF dialling

DTMF to LI and RO (AI)

Low

`TI'

Handsfree

Not connected

Low

`M3/M4'

`RI/STB'

`LSI'

Handsfree/pulse dial

Not connected

Low

Handsfree/DTMF dial

DTMF to LI and RO (AI)

Low

`TI'

`LSI'

Loudhearing

Not connected

Low

`M1/M2'

`RI/STB'

`LSI'

Loudhearing/pulse dial

Not connected

Low

Loudhearing/DTMF dial

DTMF to LI and RO (AI)

Low

`TI'

`LSI'

TAM without LSP

Signal to LI (AI)

Low

`RI/STB'

TAM with LSP

Signal to LI (AI)

Low

`RI/STB'

`LSI'

Melody feedback

PDM signal to LO (DI)

Low

`RI/STB'

Test mode 1

Reserved for testing

Test mode 2

Reserved for testing

Table 1: Operating Modes

Preliminary

AS2520/21/20B/21B

Rev. 5.1

Page 4

May 1999

Functional Description

The AS252x contains all the voice circuits needed in
a high feature telephone instrument, i .e.:

�

line adaptation (ac impedance, dc characteristics,
2/4-wire conversion, power extraction)

�

handset speech circuit

�

loudhearing with enhanced anti-Larsen

�

handsfree with dynamic loop gain control

�

switching converter

The line adaptation includes line driver, ac
impedance (return loss), 2 to 4 wire converter, dc
mask and power extraction circuit for extracting the
maximum dc power from the line to supply the whole
device and peripheral circuits.

The handset speech circuit consists of a transmit and
a receive path with mute, dual soft clipping and line
regulation (pin option). A volume control is provided
with squelch and loop gain compensation to improve
signal-to-noise ratio and to assure acoustic stability.

Loudhearing and handsfree functions are also
provided. The loudhearing function includes an anti-
Larsen circuit to prevent acoustic howling.

The handsfree circuit has a novel voice control
system which is virtually independent of any
background noise and works in a dynamic half
duplex mode as close to full duplex as the acoustic
loop gain allows.

The switching converter is used to extract the
available power from the ring signal and provides a
4V supply voltage. This allows the same loudspeaker
to be used for loudhearing/handsfree and tone
ringing.

LINE DRIVER

IMPEDANCE

SYNTESIZER

POWER

EXTRACTION

CONTROL

VOICE

POWER

CONTROL

TX-AGC

RX-AGC

GND

RTH

Vss

STB

GND

MT or PD

L +

AS2520/21/20B/21B

LEVEL

DETECTOR

300

SWITCHING

CONVERTER

Vss

LSI

VOL

LOGIC

INTERFACE

LLC

LEVEL

DETECTOR

ST-AMP

RO-AMP

LO-AMP

MI-AMP

PDM

INPUT

RING

Figure 2: Block Diagramme

Preliminary

AS2520/21/20B/21B

Rev. 5.1

Page 5

May 1999

DC Conditions

The normal operating range (off-hook) is from 13 mA
to 100 mA. Operating range with reduced
performance is from 5 mA to 13 mA (parallel
operation). In the normal operating range all
functions are operational.

In the line hold range from 0 to 5 mA the device is in
a power down mode and the voltage at LI is reduced
to maximum 3.5V.

The dc characteristic (excluding diode bridge) is
determined by the voltage at LI and the resistor R1 at
line currents above 13 mA as follows:

= V

+ I

LINE

z R1

The voltage at LI is 4.5V.

Below 13 mA the AS252x provides an additional
slope in order to allow parallel operation (see figure
3).

100

(V)

VLS

VLI

(mA)

Line Current

Typically
No ac signals
Tamb: 25癈

Figure 3: DC Mask

When the PD pin is high (during pulse dialling) the
speech circuit and other part of the device not
operating are in a power down mode to save current.
The CS pin is pulled to V

in order to turn the

external shunt transistor on to keep a low voltage
drop at the LS pin during make periods.

AC Impedance

The synthesised ac impedance of the circuit is set on
chip and by an external resistor and an external
capacitor (for complex impedance).

When R1 is set to 30 ohm, the ac impedance is 1000
ohm real, and the complex part can be set by a
capacitor connected to pin 2 (CI).

For 600 ohm telephones it is recommended to
connect a resistor and a capacitor from pin LS to V

(see application notes). The dc resistance of R1
should be kept at 30 ohm to ensure correct dc
condition.

Return loss and sidetone cancellation can be
determined independent of each other (see figure 4).

Speech Circuit

The speech circuit consists of a transmit and a
receive path with soft clipping, mute, line loss
compensation and sidetone cancellation.

Transmit

The gain of the transmit path is 36.5 dB in handset
mode (from M1/M2 to LS) and 46.5 dB in handsfree
mode (from M3/M4 to LS). The microphone inputs
have an input impedance of 15 kohm.

The unique dual soft clipping control circuit limits the
output voltage at LI to 2V

PEAK

. Dual means that the soft

clipping incorporates both a very fast control circuit to
eliminate harsh sidetone distortion and a slower
regulation circuit to limit the output voltage at 2V

PEAK

independent of the line impedance. The attack time
is 30 祍/6 dB. The overdrive range is 30 dB. When
mute is active, pin MT high, the gain is reduced by >
60 dB.

Receive

The gain of the receive path is 3 dB (test circuit
figure 8) from RI to RO. The receive input is the
differential signal of RI and STB. Also the receive
channel provides soft clipping to avoid acoustic
shock and harsh distortion.

When mute is active during dialling the gain is
reduced by > 60 dB. During DTMF dialling a MF
comfort tone is applied to the receiver. The comfort
tone is the DTMF signal with a level that is -30 dB
relative to the line signal.

Volume Control

On the AS2520 the receive gain can be changed by
pressing the volume keys. The + key increases the
gain by 10 dB in 5 steps and the � key decreases the
gain by 10 dB in 5 steps. The gain is reset by next
off-hook. The volume can also be controlled via a
microcontroller.

The AS2521 uses a potentiometer to control the
receive gain. The volume is an indirect dc control to
avoid that noise is introduced from the potentiometer.

The volume control is common for both the earpiece
and the loudspeaker. Any increase will be
compensated to ensure acoustic stability.

Preliminary

AS2520/21/20B/21B

Rev. 5.1

Page 6

May 1999

The acoustic stability is provided as follows:

When the volume is increased, e.g. by 10 dB, the
receive gain maintains the same as long as no
receive signal is applied. Applying a receive signal
will cause a 10 dB increase of the receive gain and a
corresponding decrease of the transmit gain. This
squelch function improves the signal-to-noise ratio.

In other words, a certain increase of the volume
introduces a similar amount of dynamic voice
switching, controlled by the receive signal, also in the
handset mode.

Sidetone

A good sidetone cancellation is achieved by using
the following equation:

BAL

LINE

= R5/R1

The sidetone cancellation signal is applied to the
STB input.

By using two separate Wheatstone Bridges for return
loss and sidetone cancellation it is very easy to
calculate the sidetone balance network (see figure 4).
This unique configuration provides a sidetone
cancellation less sensitive to tolerances on the
external balance network and totally independent of
the ac impedance and its tolerances.

A good and stable sidetone cancellation improves the
handsfree function considerably and ensures a safe
margin against acoustic instability under all
circumstances.

LINE

R1
30 ohm

R5
300 ohm

BAL

Figure 4: Sidetone Bridge

Furthermore, the dual Wheatstone bridge makes it
very simple to adapt the circuit to different PTT
requirements as these two parameters (return loss
and sidetone balance) are independent of each
other.

Line Loss Compensation

The line loss compensation (Rx and Tx AGC
controlled by the line current) is a pin option. When it
is activated, the transmit and receive gains are
changed by -6 dB in 1 dB steps at line currents from

20 to 50 mA or 45 to 75 mA depending on selected
range.

Loudhearing

The loudhearing mode is enabled when HS and LE
are high. In order to prevent acoustic coupling
between the handset microphone and the
loudspeaker, the AS252x incorporate an anti-Larsen
circuit.

The anti-Larsen circuit decreases the gain of the
loudspeaker amplifier when a microphone signal is
applied. If no signal is applied from the microphone,
the loudspeaker amplifier is at its full gain.

Anti-Clipping (not AS2520B/21B)

The anti-clipping circuit is activated in loudhearing
and handsfree mode. The circuit prevents harsh
distortion at very high signal levels.

Furthermore, the circuit assures that the integrity of
the whole telephone circuit is maintained under
extreme load conditions, since it prevents that the
supply voltage drops below a certain minimum level.

The attack time is fast (120 祍/6 dB) for preventing
harsh distortion when the amplitude rapidly
increases. For avoiding chopper effects and to
assure low distortion, the decay time is longer,
approx. 128 ms/6 dB.

When the anti-clipping circuit has been activated by a
large receive signal, the channel control will increase
the Tx gain corresponding to the reduction in Rx gain
caused by the anti-clipping.

Handsfree

The handsfree function allows voice communication
without using the handset (full 2-way speaker phone).
Two voice controlled attenuators prevent acoustic
coupling between the loudspeaker and the
microphone.

A conventional voice switching circuit has a channel
control with three states, namely idle, transmit or
receive. In idle state, when no signal is applied, both
the transmit and the receive channels are attenuated
by approx. 20 dB to keep the total loop gain below 0
dB.

When a signal is applied to the microphone, the
circuit switches to transmit state, i.e. the gain in the
transmit channel is increased and the gain in the
receive channel is decreased accordingly. And vice
versa when a receive signal is applied.

Preliminary

AS2520/21/20B/21B

Rev. 5.1

Page 7

May 1999

This approach has some disadvantages. It requires a
high degree of discipline, since the three state
channel control gives a very distinct half duplex with
a relative high switching time constant to avoid
chopper effects. Furthermore, the system is very
sensitive to the environment,- noise, line conditions
and acoustics (echo).

Apart from keeping a distinct discipline, the user can
not do anything to minimise the effect of these
constraints, since the parameters of the voice
switching (thresholds, time constants, noise
discrimination, etc.) can not be changed or adapted
to the actual conditions by the user.

The dynamic voice control system of the AS252x
have been designed to overcome the above
constraints. The basic philosophy behind the AS252x
is that telephone circuits should not have any
automatic regulations preventing the user from
having all information about the actual conditions
which should enable her/him to act accordingly, i.e.
to comply with the given constraints.

Now, assuming subscriber A has a handsfree
telephone and is calling subscriber B, who has a
normal telephone. The B subscriber does not
necessarily know that A is using a handsfree
telephone and will therefore not automatically comply
to the discipline of a half duplex conversation. Hence,
the disadvantages by using half duplex should apply
to the A subscriber only.

Secondly, if A is in a noisy environment, the B
subscriber should hear it, so that he speaks up to
increase the signal-to-noise ratio at the A subscriber.

The traditional 3-state switching system has two
major drawbacks: first of all, when no one is talking,
the circuit is in idle state and the B subscriber gets
the feeling that the line is dead, since the background
noise does not activate the voice switching.
Secondly, the B subscriber does not speak up, since
she/he does not hear the background noise.

The concept of the AS252x, however, does not
exclude the human factor, but provides the
information about the actual conditions to the user
and allows her/him to act accordingly, i.e. to speak
up, to change the volume, etc.

In more technical terms, the AS252x works in the
following manner:

When no signal is applied neither from the line nor
from the microphone, the circuit is in the only static
state, which is transmit channel full open and receive
channel attenuated by up to 30 dB.

The advantages of using the transmit state as the
static (idle) state are that the B subscriber hears an
open line (the line is not dead), does not miss the
initial word of a sentence when the A subscriber
starts talking, and hears the level of the background
noise at A磗 end which will actuate her/him to speak
up accordingly.

When the A subscriber starts talking, the circuit
remains in the static state.

The dynamic state of the voice switching can only be
activated by the receive signal. Applying a receive
signal above a certain level will cause the circuit to
enter the dynamic state.

VOL

I
D

LINE

� 10 dB

AGC

PEAK

DETECTOR

2/4

Figure 5: Channel Control System

The signal for controlling the channel attenuation is
taken after the sidetone amplifier. With the volume at
0 dB (neutral) the threshold for entering the dynamic
state (V

) is 15 mV assuming that V

> V

(see

figure 5).

In the dynamic state the channel attenuation is
controlled by a voltage controlled amplifier. The
attack time is 4 ms/6 dB and the hold time is 200 ms.

A speech compression is activated when a transmit
signal with a high amplitude reaches a level
corresponding to approximately 460 mV on the line.

1.50

1.25

1.00

0.75

0.50

0.25

0.00

100

150

200

250

300

Microphone Input Signal

i
n

t
p

t

S

i
g

(mV)

Sidetone Cancellation: 11 dB
Volume Control: 0 dB (neutral)

Figure 6: Speech Compression

Preliminary

AS2520/21/20B/21B

Rev. 5.1

Page 8

May 1999

The speech compression allows a higher gain in the
transmit channel, i.e. the microphone gets more
sensitive at low sound pressure levels on the
microphone, which enables the user to move further
away from the telephone. This means that a constant
signal is provided on the line practical independent of
the microphone signal level. Any reduction of gain by
the compressor in the transmit channel will
automatically be given to the receive channel.

Switching Converter

The ac ringing signal is utilised to extract the power
necessary to the tone ringer circuit. A switch mode
power supply is used to obtain a high efficiency dc
conversion.

This approach allows the use of the same
loudspeaker and amplifiers for both loudhearing and
tone ringing. It also allows an acoustic feedback of
the melodies during programming with the same
sound pressure level as during ringing.

When a ringing signal is applied, PD is pulled high
and the oscillator is enabled. The switching converter
is controlled by the output CM, which is turned high
and low with a duty cycle controlled by the voltage at
V

When off-hook the switching converter has a high
impedance (CM low) to avoid any influence on the
transmission and on pulse dialling.

The smoothing capacitor should be in the range of 10
to 68 nF. The choke coil must have an inductance of
>1mH and a dc resistance of < 15 ohm.

Vss

2.2 mH

327

470 �

1�5

510

5k6

33 n

10 k

5V1

547

30V

Figure 7: Switching Converter

Tone Input

The tone input is a digital input in ringing mode and
during melody feedback. The digital melody signal
(PDM = pulse density modulation) is directly applied
to the TI input (see also application notes for further
details).

During DTMF dialling the DTMF signal is applied
through a capacitor to the TI input and will be fed to
the line (pin LI) and to the receive output (RO) as
confidence tone.

Preliminary

AS2520/21/20B/21B

Rev. 5.1

Page 9

May 1999

Electrical Characteristics

Absolute Maximum Ratings*

Supply Voltage............................................................................................................................... -0.3

Input Current..........................................................................................................................................+/- 25 mA

Input Voltage (LS) ....................................................................................................................... -0.3V

10V

Input Voltage (LI, CS, SS).............................................................................................................-0.3V

Input Voltage (STB, RI) .........................................................................................................-2V

+0.3V

Digital Input Voltage .......................................................................................................... -0.3V

+ 0.3V

Electrostatic Discharge ..........................................................................................................................+/- 1000V

Storage Temperature Range........................................................................................................... -65 to +125癈

Total Power Dissipation ............................................................................................................................ 500mW

*Exceeding these figures may cause permanent damage. Functional operation under these conditions is not permitted.

Recommended Operating Range

Symbol

Parameter

Conditions

Min.

Typ.*

Max.

Units

Supply Voltage (internally generated)

Speech mode

3.0

4.1

5.5

Supply Voltage (internally regulated)

Speech mode

3.0

4.1

5.5

AMB

Ambient Operating Temp. Range

-25

+70

癈

* Typical figures are at 25癈 and are for design aid only; not guaranteed and not subject to production testing.

DC Characteristics (I

LINE

= 15 mA, recommended operating conditions unless otherwise specified)

Symbol

Parameter

Conditions

Min.

Typ.

Max.

Units

Operating Supply Current

HS = high

LE = high

HS and LE = high

PD = high, CM running

300

礎

DDPD

DD0

Power-Down Current

Standby Current

PD = high

All digital inputs = V

200

礎

Line Voltage

13 mA< I

LINE

< 100 mA

4.2

4.5

4.8

Output Current, Sink
Pin CS, SS

= 0.4V

1.5

Output Current, Sink
Pin CM

= 0.4V

1.5

Input Low Voltage

AMB

= 25癈

0.2 V

Input High Voltage

AMB

= 25癈

0.8 V

Preliminary

AS2520/21/20B/21B

Rev. 5.1

Page 10

May 1999

AC Electrical Characteristics

LINE

= 15 mA; f = 800 Hz; recommended operating conditions unless otherwise specified.

Transmit

Symbol

Parameter

Conditions

Min.

Typ.

Max.

Units

Gain (M1/M2 to LS)

Gain (M3/M4 to LS)

HS, LH modes; LLC = A

GND

HF mode; LLC = A

GND

36.5

46.5

Gain (TI to LS)

MF mode

13.5

TX/F

Variation with Frequency

f = 500 Hz to 3.4 kHz

+/- 0.8

LLC

Gain Range, LLC

Speech mode; LLC = V

or V

-6

THD

Distortion

< 0.25 V

RMS

AGC

SCO

Soft Clip Level

Soft Clip Overdrive

HS, LH modes; V

HF mode; V

650

PEAK

Input Impedance;

M1/M2 and M3/M4

kohm

Attenuation Depth

MUTE

Mute Attenuation

Mute activated

Noise Output Voltage

HS = high; T

AMB

= 25癈

LE = high; HS = low; T

AMB

= 25癈

-72

-62

dBmp

IN MAX

Input Voltage Range;
M1/M2

Differential

Single ended

+/- 1

+/- 0.5

PEAK

Line Driver

Symbol

Parameter

Test Conditions

Min.

Typ.

Max.

Units

IN MAX

Input Voltage Range; LI

+/- 2

PEAK

AC/TEMP

Return Loss

Temperature Variation

= 1000 ohm; T

AMB

= 25癈

0.5

/癈

Preliminary

AS2520/21/20B/21B

Rev. 5.1

Page 11

May 1999

Receive

Symbol

Parameter

Condition

Min.

Typ.

Max.

Units

Gain (LS to RO), Default

LSP Gain (LSI to LO)

Volume reset

1.5

17.5

4.5

20.5

TX/F

Variation with Frequency

f = 500 Hz to 3.4 kHz

+/- 0.8

LLC

Gain Range, LLC

Speech mode; LLC = V

or V

-6

Volume Range

10 steps, each 2 dB

THD

Distortion

< 0.2 V

RMS

SCO

Soft Clip Level (RO)

Soft Clip Level (LO)

Soft Clip Overdrive

Not AS2520B/21B; V

Unloaded

1.3

PEAK

RTH

DECAY

Threshold Voltage at RTH

Attenuation Depth

Attack Time

Decay Time

Channel control; V

> 0.8 V

RMS

Channel control

祍/6dB

UFC

Noise Output Voltage (RO)

Unwanted Frequency
Components (RO)

HS = high; T

AMB

= 25癈

50 Hz.........20 kHz

-72

-60

dBmp

Input Impedance, RI

kohm

IN RI

Input Voltage Range, RI

+/- 2

PEAK

Sidetone Cancellation

< 0.2 V

RMS

; T

AMB

= 25癈

Input Impedance, STB

kohm

IN ST

Input Voltage Range, STB

+/- 2

PEAK

General Timings

Symbol

Parameter

Condition

Min.

Typ.

Max.

Units

VOL

Volume Key Debounce

SCA

Soft Clip Attack Time

above soft clip level

0.12

ms/6dB

SCD

Soft Clip Decay Time

below soft clip level

128

ms/6dB

PDA

Peak Detector Attack Time V

above V

3.2

ms/V

PDD

Peak Detector Decay Time V

below V

ms/V

LPA

Low-Power Attack Time

< 3.6V

250

ms/6dB

LPD

Low-Power Release

> 3.6V

sec/6dB

Preliminary

AS2520/21/20B/21B

Rev. 5.1

Page 13

May 1999

Application Diagramme

I
N

/
T

I
C

I
R

I
T

Vss

STB

LSI

GND

RTH

SSP

LLC

15 n

10 n

300

1k8

7k5

10 �

100 n

100 �

470 �

5k6

10 k

2N5551

2.2 mH

5V1

MPSA92

33 n

510

1�5

510

1�5

BC327

100 �

33 n

10 �

BC327

10 V

VOL

100 k

Side tone
balance
network

GND

15 n

100 n

2k2

1k2

220 �

1 �

INPUT FOR DTMF

AND

TONE RINGER MELODIES

CONTROL INPUTS

(FROM 礐)

VOL+

VOL-

10 k

AS2521

AS2520

High

Low

Off

1k8

Figure 9: Application Diagramme

Applications Hints

Interface to Microcontroller

In off-hook condition the microcontroller can be supplied from V

of AS252x. The digital inputs (HS, LE, PD, and

MT) must be kept low until V

has reached its minimum operating voltage (>2.5V).

Radio Frequency Interference

The RFI sensitivity has been minimised by the consequent use of CMOS technology and one overall ground and
by having differential inputs with a relative low input impedance.

For further application information see application notes for the AS2520 series.

Preliminary

AS2520/21/20B/21B

Rev. 5.1

Page 14

May 1999

Pin Configuration

28 Pin SOP/DIP

GND

STB

LLC

RTH

LSI

SSP

VOL

Ordering Information

Part
Number

Package
Type

Volume
Control

Soft Clip
Loudspk.

AS2520 T

28 pin
SOP

Digital

Yes

AS2520 P

28 pin DIP

Digital

Yes

AS2520B T

28 pin
SOP

Digital

AS2520B
P

28 pin DIP

Digital

AS2521 T

28 pin
SOP

Analogue

Yes

AS2521 P

28 pin DIP

Analogue

Yes

AS2521B T

28 pin
SOP

Analogue

AS2521B
P

28 pin DIP

Analogue

The devices are also available as dice on request.

Devices sold by Austria Mikro Systeme Int. AG are covered by the warranty and patent indemnification provisions appearing in
its Term of Sale. Austria Mikro Systeme Int. AG makes no warranty, express, statutory, implied, or by description regarding
the information set forth herein or regarding the freedom of the described devices from patent infringement Austria Mikro
Systeme Int. AG reserves the right to change specifications and prices at any time and without notice. Therefore, prior to
designing this product into a system, it is necessary to check with Austria Mikro Systeme Int. AG for current information. This
product is intended for use in normal commercial applications. Applications requiring extended temperature range, unusual
environmental requirements, or high reliability applications, such as military, medical life-support or life-sustaining equipment
are specifically not recommended without additional processing by Austria Mikro Systeme Int. AG for each application.

Copyright � 1999, Austria Mikro Systeme International AG, Schloss Premst鋞ten, 8141 Unterpremst鋞ten, Austria.
Trademarks Registered�. All rights reserved. The material herein may not be reproduced, adapted, merged, translated,
stored, or used without the prior written consent of the copyright owner.

Austria Mikro Systeme Int. AG reserves the right to change or discontinue this product without notice.

协谢械泻褌褉芯薪薪褘泄 泻芯屑锌芯薪械薪褌: AS2520

协谢械泻褌褉芯薪薪褘泄泻芯屑锌芯薪械薪褌: AS2520